date:20130918

Re: [asterisk-users] RTP port ranges

2013-09-18 Thread Ron Wheeler


I only use 100 ports as well but we have a very low call volume.
I thought that I saw that you need to allocate 2 ports for every 
simultaneous call that you need to support.
The ports are free (no charge) and are UDP not TCP so you do not lose 
any TCP ports.


I am not sure what a hacker could do if they attacked these ports.

Ron

On 18/09/2013 2:29 PM, Ira wrote:

Re: [asterisk-users] RTP port ranges Hello Thorsten,

Tuesday, September 17, 2013, 1:05:15 AM, you wrote:


Where is it stated that you MUST use 1-2 ???

Someone else please ?



Well, I don't use that range. This is that part of my rtp.conf

rtpstart=16000
rtpend=16100

I knew I didn't need the default 25000 ports, in fact 100 is probably 
more than 10 times what I'll ever need.


Been working for for 5 years with those numbers. I decided when I 
first did this that if I used non standard ports I might be less 
susceptible to hacking. Probably not accurate, but I did it anyway.


-- Ira


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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] sipgate outgoing calls

2013-09-18 Thread David Duffett

I believe registration is in place, otherwise inbound calls would not work.

Also, registration is not required for outbound calls to work.

I would suggest cutting down your sip.conf profile to this minimal
configuration:

host=sipgate.co.uk
username=xxx
fromuser=xxx
insecure=invite,port
secret=xxx
context=my-inbound-context
type=peer

If outbound calls still do not with this, I would suggest that there may be
an issue in the general section of your sip.conf

Assuming calls do work, you can then add any other configuration lines you
feel are necessary - but remember, as with all Asterisk configuration
files, less is more :-)
 On 18 Sep 2013 22:06, "Administrator TOOTAI"  wrote:

> Le 18/09/2013 15:29, gpxctawjc...@irational.org a écrit :
>
>> Hello
>>
>
> Hi
>
>
>> i am trying to setup sipgate gateway
>>
>> i can get incoming calls fine, but when i dial in and then try to dial
>> out i get this in asterisk command line
>>
>> -- Called 01179248615@sipgate
>> [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
>> handle_response_invite: Failed to authenticate on INVITE to
>> '"01179553708" ;**tag=as30eb9dd1'
>> -- SIP/sipgate-014d is circuit-busy
>>   == Everyone is busy/congested at this time (1:0/1/0)
>>
>>
>> here is my sip.conf file
>>
>>
>> [general]
>> port = 5060
>> bindaddr = 0.0.0.0
>> context=default
>> qualify=no
>> disallow=all
>> allow=alaw
>> allow=ulaw
>> allow=g729
>> allow=gsm
>> allow=slinear
>> srvlookup=yes
>> videosupport=yes
>> alwaysauthreject=yes
>>
>> register => 
>> SIP-ID:SIP-Password@sipgate.**co.uk/SIP-ID
>>
>> [sipgate]
>> type=peer
>> secret=SIP_PASSWORD
>> insecure=invite
>> username=SIP-ID
>> defaultuser=SIP-ID
>> fromuser=SIP-ID
>> context=sipgate_in
>> fromdomain=sipgate.co.uk
>> host=sipgate.co.uk
>> outboundproxy=proxy.live.**sipgate.co.uk
>> qualify=yes
>> disallow=all
>> allow=alaw
>> dtmfmode=rfc2833
>>
>>
>> SIP-ID:SIP-Password
>> obviously, i replace these with my login details
>>
>> but, are these the same thing ?
>> SIP-Password
>> SIP_PASSWORD
>>
>> the sipgate guides are contradictory
>>
>> http://www.sipgate.com/faq/**article/394/How_do_I_**configure_Asterisk
>> http://www.live.sipgate.co.uk/**faq/article/508/How_do_I_**
>> configure_Asteri
>> sk
>>
>>
>> any suggestions ?
>>
>> Many thanks
>>
>
> My setup with sipgate.de
>
> [sipgate]
> type=peer
> secret=MY-PASSWORD
> defaultuser=SIP-ID
> host=217.10.79.9
> fromuser=SIP-ID
> fromdomain=sipgate.de
> context=incoming-sipgate
> ;qualify=900
> dtmfmode=info
> directmedia=yes
> insecure=port,invite
> disallow=all
> allow=ulaw,alaw
> accountcode=MY-ACCOUNTCODE
>
> What you forget is to register with them:
>
> ; Sipgate
> register => 
> SIP-ID:my-passw...@sipgate.de/**SIP-ID;don't
>  accept to register without FQDN
>
> Hope that help
>
> --
> Daniel
>
> --
> __**__**_
> -- 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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Re: [asterisk-users] sipgate outgoing calls

2013-09-18 Thread Administrator TOOTAI


Le 18/09/2013 15:29, gpxctawjc...@irational.org a écrit :

Hello


Hi



i am trying to setup sipgate gateway

i can get incoming calls fine, but when i dial in and then try to dial
out i get this in asterisk command line

-- Called 01179248615@sipgate
[Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
handle_response_invite: Failed to authenticate on INVITE to
'"01179553708" ;tag=as30eb9dd1'
-- SIP/sipgate-014d is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)


here is my sip.conf file


[general]
port = 5060
bindaddr = 0.0.0.0
context=default
qualify=no
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
videosupport=yes
alwaysauthreject=yes

register => SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID

[sipgate]
type=peer
secret=SIP_PASSWORD
insecure=invite
username=SIP-ID
defaultuser=SIP-ID
fromuser=SIP-ID
context=sipgate_in
fromdomain=sipgate.co.uk
host=sipgate.co.uk
outboundproxy=proxy.live.sipgate.co.uk
qualify=yes
disallow=all
allow=alaw
dtmfmode=rfc2833


SIP-ID:SIP-Password
obviously, i replace these with my login details

but, are these the same thing ?
SIP-Password
SIP_PASSWORD

the sipgate guides are contradictory

http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk
http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri
sk


any suggestions ?

Many thanks


My setup with sipgate.de

[sipgate]
type=peer
secret=MY-PASSWORD
defaultuser=SIP-ID
host=217.10.79.9
fromuser=SIP-ID
fromdomain=sipgate.de
context=incoming-sipgate
;qualify=900
dtmfmode=info
directmedia=yes
insecure=port,invite
disallow=all
allow=ulaw,alaw
accountcode=MY-ACCOUNTCODE

What you forget is to register with them:

; Sipgate
register => SIP-ID:my-passw...@sipgate.de/SIP-ID ;don't accept to 
register without FQDN


Hope that help

--
Daniel

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Re: [asterisk-users] RTP port ranges

2013-09-18 Thread Ira

Hello Thorsten,

Tuesday, September 17, 2013, 1:05:15 AM, you wrote:


Where is it stated that you MUST use 1-2 ???

Someone else please ?



Well, I don't use that range. This is that part of my rtp.conf

rtpstart=16000
rtpend=16100

I knew I didn't need the default 25000 ports, in fact 100 is probably more than 
10 times what I'll ever need.

Been working for for 5 years with those numbers. I decided when I first did 
this that if I used non standard ports I might be less susceptible to hacking. 
Probably not accurate, but I did it anyway.

-- Ira--
_
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[asterisk-users] sipgate outgoing calls

2013-09-18 Thread gpxctawjc5oh


Hello

i am trying to setup sipgate gateway

i can get incoming calls fine, but when i dial in and then try to dial
out i get this in asterisk command line

-- Called 01179248615@sipgate
[Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
handle_response_invite: Failed to authenticate on INVITE to
'"01179553708" ;tag=as30eb9dd1'
-- SIP/sipgate-014d is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)


here is my sip.conf file


[general]
port = 5060
bindaddr = 0.0.0.0
context=default
qualify=no
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
videosupport=yes
alwaysauthreject=yes

register => SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID

[sipgate]
type=peer
secret=SIP_PASSWORD
insecure=invite
username=SIP-ID
defaultuser=SIP-ID
fromuser=SIP-ID
context=sipgate_in
fromdomain=sipgate.co.uk
host=sipgate.co.uk
outboundproxy=proxy.live.sipgate.co.uk
qualify=yes
disallow=all
allow=alaw
dtmfmode=rfc2833


SIP-ID:SIP-Password
obviously, i replace these with my login details

but, are these the same thing ?
SIP-Password
SIP_PASSWORD

the sipgate guides are contradictory

http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk
http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri
sk


any suggestions ?

Many thanks


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Re: [asterisk-users] Can't connect to Asterisk cli

2013-09-18 Thread Asghar Mohammad

i think you messed 2 installs of asterisk.
if you compile asterisk from sources it not insert init script.
you can test installing to /opt.

1. cd to asterisk sources folder
2. make distclean
3. ./configure --prefix=/opt/asterisk
4. make
5. sudo make install
6. /opt/asterisk/sbin/asterisk -c
you can remove this installation by sudo rm -rfv /opt/asterisk
if it work then you should remove every asterisk installation and then
install a fresh copy .(or reinstall OS if you cannot remove)
hope this will help you.



On Wed, Sep 18, 2013 at 2:57 PM, Asmaa Ahmed  wrote:

> It looks this is because Asterisk isn't started
> when tried to start it, I got a core dump!
> $ /etc/init.d/asterisk start
>  * Starting Asterisk PBX: asterisk
> Illegal instruction (core dumped)
>
> --
> From: asabatg...@hotmail.com
> To: asterisk-users@lists.digium.com
> Subject: Can't connect to Asterisk cli
> Date: Wed, 18 Sep 2013 14:13:16 +0200
>
> Hello,
>
> I have started using Asterisk recently on my Ubuntu server. I installed it
> first using apt-get and it worked fine sort of, but still couldn't hear
> voice during the call!
> I read that this problem solved by reinstalling it, so I decided to
> reinstall the latest version from the source as apt-get can give you only
> Asterisk 1.8
> So I have installed Asterisk 11 following the procedure in Asterisk- The
> Definitive Guide, 4th Edition book
> The installation done with no problem, but now I can't even login to CLI,
> it keeps returns Linux prompt!
> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r
> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc
> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv
> ubuntu@vbefe01:/etc/asterisk$
>
> and if I don't use sudo, it returns a core dump
> $ asterisk -vvc
> Illegal instruction (core dumped)
>
> I was able to connect before installing the latest version from the
> source? How can I fix that?
>
> Thanks.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Can't connect to Asterisk cli

2013-09-18 Thread Ishfaq Malik

Ah, my bad, possibly not.


On 18 September 2013 13:50, Asghar Mohammad  wrote:

> SELinux  exists in Ubuntu?
>
>
>
> On Wed, Sep 18, 2013 at 2:45 PM, Ishfaq Malik  wrote:
>
>> Have you checked your SELinux settings?
>>
>>
>> On 18 September 2013 13:13, Asmaa Ahmed  wrote:
>>
>>> Hello,
>>>
>>> I have started using Asterisk recently on my Ubuntu server. I installed
>>> it first using apt-get and it worked fine sort of, but still couldn't hear
>>> voice during the call!
>>> I read that this problem solved by reinstalling it, so I decided to
>>> reinstall the latest version from the source as apt-get can give you only
>>> Asterisk 1.8
>>> So I have installed Asterisk 11 following the procedure in Asterisk- The
>>> Definitive Guide, 4th Edition book
>>> The installation done with no problem, but now I can't even login to
>>> CLI, it keeps returns Linux prompt!
>>> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r
>>> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc
>>> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv
>>> ubuntu@vbefe01:/etc/asterisk$
>>>
>>> and if I don't use sudo, it returns a core dump
>>> $ asterisk -vvc
>>> Illegal instruction (core dumped)
>>>
>>> I was able to connect before installing the latest version from the
>>> source? How can I fix that?
>>>
>>> Thanks.
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> --
>>
>> Ishfaq Malik
>> Department: VOIP Support
>> Company: Packnet Limited
>> t: +44 (0)845 004 4994
>> f: +44 (0)161 660 9825
>> e: i...@pack-net.co.uk
>> w: http://www.pack-net.co.uk
>>
>> Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
>> 37 Ducie Street
>> Manchester, M1 2JW
>> COMPANY REG NO. 04920552
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

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

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
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Re: [asterisk-users] Can't connect to Asterisk cli

2013-09-18 Thread Asmaa Ahmed

It looks this is because Asterisk isn't startedwhen tried to start it, I got a 
core dump!$ /etc/init.d/asterisk start * Starting Asterisk PBX: asteriskIllegal 
instruction (core dumped)
From: asabatg...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: Can't connect to Asterisk cli
Date: Wed, 18 Sep 2013 14:13:16 +0200




Hello,
I have started using Asterisk recently on my Ubuntu server. I installed it 
first using apt-get and it worked fine sort of, but still couldn't hear voice 
during the call!
I read that this problem solved by reinstalling it, so I decided to reinstall 
the latest version from the source as apt-get can give you only Asterisk 1.8So 
I have installed Asterisk 11 following the procedure in Asterisk- The 
Definitive Guide, 4th Edition bookThe installation done with no problem, but 
now I can't even login to CLI, it keeps returns Linux prompt! 
ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -rubuntu@vbefe01:/etc/asterisk$ 
sudo asterisk -vcubuntu@vbefe01:/etc/asterisk$ sudo asterisk 
-cvvvubuntu@vbefe01:/etc/asterisk$   
and if I don't use sudo, it returns a core dump$ asterisk -vvcIllegal 
instruction (core dumped)
I was able to connect before installing the latest version from the source? How 
can I fix that?
Thanks. 
  --
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Re: [asterisk-users] Can't connect to Asterisk cli

2013-09-18 Thread Asghar Mohammad

SELinux  exists in Ubuntu?



On Wed, Sep 18, 2013 at 2:45 PM, Ishfaq Malik  wrote:

> Have you checked your SELinux settings?
>
>
> On 18 September 2013 13:13, Asmaa Ahmed  wrote:
>
>> Hello,
>>
>> I have started using Asterisk recently on my Ubuntu server. I installed
>> it first using apt-get and it worked fine sort of, but still couldn't hear
>> voice during the call!
>> I read that this problem solved by reinstalling it, so I decided to
>> reinstall the latest version from the source as apt-get can give you only
>> Asterisk 1.8
>> So I have installed Asterisk 11 following the procedure in Asterisk- The
>> Definitive Guide, 4th Edition book
>> The installation done with no problem, but now I can't even login to CLI,
>> it keeps returns Linux prompt!
>> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r
>> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc
>> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv
>> ubuntu@vbefe01:/etc/asterisk$
>>
>> and if I don't use sudo, it returns a core dump
>> $ asterisk -vvc
>> Illegal instruction (core dumped)
>>
>> I was able to connect before installing the latest version from the
>> source? How can I fix that?
>>
>> Thanks.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
> Ishfaq Malik
> Department: VOIP Support
> Company: Packnet Limited
> t: +44 (0)845 004 4994
> f: +44 (0)161 660 9825
> e: i...@pack-net.co.uk
> w: http://www.pack-net.co.uk
>
> Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
> 37 Ducie Street
> Manchester, M1 2JW
> COMPANY REG NO. 04920552
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Can't connect to Asterisk cli

2013-09-18 Thread Ishfaq Malik

Have you checked your SELinux settings?


On 18 September 2013 13:13, Asmaa Ahmed  wrote:

> Hello,
>
> I have started using Asterisk recently on my Ubuntu server. I installed it
> first using apt-get and it worked fine sort of, but still couldn't hear
> voice during the call!
> I read that this problem solved by reinstalling it, so I decided to
> reinstall the latest version from the source as apt-get can give you only
> Asterisk 1.8
> So I have installed Asterisk 11 following the procedure in Asterisk- The
> Definitive Guide, 4th Edition book
> The installation done with no problem, but now I can't even login to CLI,
> it keeps returns Linux prompt!
> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r
> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc
> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv
> ubuntu@vbefe01:/etc/asterisk$
>
> and if I don't use sudo, it returns a core dump
> $ asterisk -vvc
> Illegal instruction (core dumped)
>
> I was able to connect before installing the latest version from the
> source? How can I fix that?
>
> Thanks.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

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

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Can't connect to Asterisk cli

2013-09-18 Thread Asghar Mohammad

become root sudo su - or su -l give your password.
if asterisk is already running connect to "asterisk -rvvvc" otherwisw
"asterisk -c".
if you want asterisk run as daemon "asterisk" and then connect to asterisk
"asterisk -rvvvc"


On Wed, Sep 18, 2013 at 2:13 PM, Asmaa Ahmed  wrote:

> Hello,
>
> I have started using Asterisk recently on my Ubuntu server. I installed it
> first using apt-get and it worked fine sort of, but still couldn't hear
> voice during the call!
> I read that this problem solved by reinstalling it, so I decided to
> reinstall the latest version from the source as apt-get can give you only
> Asterisk 1.8
> So I have installed Asterisk 11 following the procedure in Asterisk- The
> Definitive Guide, 4th Edition book
> The installation done with no problem, but now I can't even login to CLI,
> it keeps returns Linux prompt!
> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r
> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc
> ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv
> ubuntu@vbefe01:/etc/asterisk$
>
> and if I don't use sudo, it returns a core dump
> $ asterisk -vvc
> Illegal instruction (core dumped)
>
> I was able to connect before installing the latest version from the
> source? How can I fix that?
>
> Thanks.
>
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[asterisk-users] Can't connect to Asterisk cli

2013-09-18 Thread Asmaa Ahmed

Hello,
I have started using Asterisk recently on my Ubuntu server. I installed it 
first using apt-get and it worked fine sort of, but still couldn't hear voice 
during the call!
I read that this problem solved by reinstalling it, so I decided to reinstall 
the latest version from the source as apt-get can give you only Asterisk 1.8So 
I have installed Asterisk 11 following the procedure in Asterisk- The 
Definitive Guide, 4th Edition bookThe installation done with no problem, but 
now I can't even login to CLI, it keeps returns Linux prompt! 
ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -rubuntu@vbefe01:/etc/asterisk$ 
sudo asterisk -vcubuntu@vbefe01:/etc/asterisk$ sudo asterisk 
-cvvvubuntu@vbefe01:/etc/asterisk$   
and if I don't use sudo, it returns a core dump$ asterisk -vvcIllegal 
instruction (core dumped)
I was able to connect before installing the latest version from the source? How 
can I fix that?
Thanks.   --
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Re: [asterisk-users] RTP not being switched between both SIP endpoints

2013-09-18 Thread Gareth Blades


On 18/09/13 12:40, Kenny Watson wrote:

Hi,

Since opensips is not handling media (i presume) is the audio not already going 
direct from asterisk to the other endpoint?

Thanks

Kenny


Opensips wasnt handling the media so the audio was between the original 
caller and asterisk (with the signalling being relayed by opensips). It 
was just when we dialled onto the final destination via SIP asterisk 
stayed in the loop and didnt issue a reinvite.


Its all fixed now. Although we weren't using any features the AGI 
application was setting DYNAMIC_FEATURES to an empty string which was 
enough to keep asterisk in a loop. We stopped the AGI from setting the 
variable if there were no features and it started working.


Thanks
Gareth

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Re: [asterisk-users] RTP not being switched between both SIP endpoints

2013-09-18 Thread Kenny Watson

Hi,

Since opensips is not handling media (i presume) is the audio not already going 
direct from asterisk to the other endpoint?

Thanks

Kenny

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] on behalf of Gareth Blades 
[mailinglist+aster...@dns99.co.uk]
Sent: 17 September 2013 11:17
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] RTP not being switched between both SIP endpoints

We have a system where calls are coming in from telcos via an opensips
server and then being redirected out to a regular sip destination.
There is no NAT, DTMF features, call recording, or codec translation
being performed so I would expect asterisk to issue a reinvite after the
call is answered and switch the audio however it is not happening.

Here is the sip peer information for the call coming from opensips.
Directmedia is not specifically defined so its using the asterisk
default value.

   * Name   : vmpubopensips3
   Description  :
   Secret   : 
   MD5Secret: 
   Remote Secret: 
   Context  : from-pubopensips
   Record On feature : automon
   Record Off feature : automon
   Subscr.Cont. : 
   Language :
   Tonezone : 
   AMA flags: Unknown
   Transfer mode: open
   CallingPres  : Presentation Allowed, Not Screened
   Callgroup:
   Pickupgroup  :
   Named Callgr :
   Nam. Pickupgr:
   MOH Suggest  :
   Mailbox  :
   VM Extension : asterisk
   LastMsgsSent : 0/0
   Call limit   : 0
   Max forwards : 0
   Dynamic  : No
   Callerid : "" <>
   MaxCallBR: 384 kbps
   Expire   : -1
   Insecure : no
   Force rport  : Auto (No)
   Symmetric RTP: No
   ACL  : No
   DirectMedACL : No
   T.38 support : No
   T.38 EC mode : Unknown
   T.38 MaxDtgrm: -1
   DirectMedia  : Yes
   PromiscRedir : No
   User=Phone   : No
   Video Support: No
   Text Support : No
   Ign SDP ver  : No
   Trust RPID   : Yes
   Send RPID: No
   Subscriptions: Yes
   Overlap dial : No
   DTMFmode : rfc2833
   Timer T1 : 500
   Timer B  : 32000
   ToHost   : 88.x.x.x
   Addr->IP : 88.x.x.x:5060
   Defaddr->IP  : (null)
   Prim.Transp. : UDP
   Allowed.Trsp : UDP
   Def. Username:
   SIP Options  : (none)
   Codecs   : (gsm|ulaw|alaw)
   Codec Order  : (alaw:20,ulaw:20,gsm:20)
   Auto-Framing :  No
   Status   : Unmonitored
   Useragent:
   Reg. Contact :
   Qualify Freq : 6 ms
   Keepalive: 0 ms
   Sess-Timers  : Accept
   Sess-Refresh : uas
   Sess-Expires : 1800 secs
   Min-Sess : 90 secs
   RTP Engine   : asterisk
   Parkinglot   :
   Use Reason   : No
   Encryption   : No

When the call comes in the SDP contains :-

v=0.
o=root 973184584 973184584 IN IP4 81.x.x.x
s=session.
c=IN IP4 81.x.x.x
t=0 0.
m=audio 11370 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

and we reply back with :-

v=0.
o=root 822402971 822402971 IN IP4 88.x.x.x
s=Asterisk PBX 11.2-cert2.
c=IN IP4 88.x.x.x
t=0 0.
m=audio 10428 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


When we send the outbound SIP information we advertise the following SDP :-

v=0.
o=root 431105643 431105643 IN IP4 88.x.x.x
s=Asterisk PBX 11.2-cert2.
c=IN IP4 88.x.x.x
t=0 0.
m=audio 10144 RTP/AVP 8 3 0 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

and the other end replies with :-

v=0.
o=hksbc1a 609621538 609621538 IN IP4 203.x.x.x
s=sip call.
c=IN IP4 203.x.x.x
t=0 0.
m=audio 34146 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=ptime:20.
a=fmtp:101 0-15.

In the Dial() command the only option we are using is M() which is used
to run a macro when the call is answered. This is used to update cdr
records and perform other features if they are enabled. In this case we
are just updating the cdr records so I would expect the audio to be
switched as soon as the macro finishes.

Any ideas what could be wrong?
We are running Asterisk PBX 11.2-cert2

Thanks
Gareth

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[asterisk-users] Can't connect to Asterisk cli

Re: [asterisk-users] RTP not being switched between both SIP endpoints

Re: [asterisk-users] RTP not being switched between both SIP endpoints

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