Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Gopalakrishnan N 
By having different server, i made it work. I suspect some network issue...


On Wed, Jul 3, 2013 at 3:27 AM, Asghar Mohammad  wrote:

> make a call and post cli log
>
>
> On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> still the peer shows unreachable let me restart and give a try...
>>
>>
>> On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad wrote:
>>
>>> *1st Location*
>>> [manila]
>>> type=peer
>>> username=indman01
>>> secret=indman01
>>> host=10.30.2.5 <-- ip of 2nd location
>>> port=5060
>>> context=Manila
>>> insecure=port,invite
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> directmedia=no
>>> qualify=yes
>>> disallow=all
>>> allow=g729
>>> allow=ulaw
>>>
>>> 1st location dialplan
>>> exten => _2XXX,1,Dial(SIP/manila/${EXTEN}
>>> )
>>> exten => _2XXX,n,Hangup
>>>
>>> *2nd Location*
>>> [india]
>>> type=friend
>>> username=manind01
>>> secret=manind01
>>> host=dynamic
>>> port=5060
>>> context=10.20.111.48 <- ip of 1st location
>>>  insecure=port,invite
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> directmedia=no
>>> qualify=yes
>>> nat=force_rport,comedia
>>> disallow=all
>>> allow=g729
>>> allow=ulaw
>>> allow=alaw
>>>
>>> 2st location dialplan
>>> exten => _2XXX,1,Dial(SIP/india/${EXTEN} 
>>> )
>>> exten => _2XXX,n,Hangup
>>>
>>> then you should handle the call when it arrive in any server
>>> let me know if it work.
>>>
>>>
>>> On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
 I tried creating two trunks with following,
 *1st Location*
 [10.30.2.5]
 type=friend
 username=indman01
 secret=indman01
 host=dynamic
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw

 *2nd Location*
 [10.20.111.48]
 type=friend
 username=manind01
 secret=manind01
 host=dynamic
 port=5060
 context=india
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw

 My dialplan is like this
 exten => _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN}
 )
 exten => _2XXX,n,Hangup

 And the output I get is
  Executing [2001@Test:1] Dial("SIP/3081-27d2", "SIP/10.30.2.5/2001")
 in new stack
 [Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
 Subscriber absent)
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Executing [2001@Test:2] Hangup("SIP/3081-27d2", "") in new
 stack
   == Spawn extension (Test, 2001, 2) exited non-zero on
 'SIP/3081-27d2'

 Actually the trunk which i mentioned in my first email, it was
 working... and from today it is not

 Still breaking... what could be the reason... !



 On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad wrote:

> yes you can. just create trunks on both side with static ip and in
> dial use trunk name.
> exten => _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =>
> _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
> make a call from a to b and one from b to and post cli log here or
> upload anyware else.
>
>
> On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> can't we use without register command both way as peer to peer?
>>
>>
>> On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad 
>> wrote:
>>
>>> 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on
>>> b and 10.10.10.0 on a.
>>> 2. use host=dynamic type=friend on  side A and host=ip type=peer on
>>> side B.
>>> 3. general section in sip.conf of side B register with server A.
>>>
>>> please see comments in sip.conf
>>> ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
>>> registering
>>> ; as any IP address used for
>>> staticly defined
>>> ; hosts.  This helps avoid the
>>> configuration
>>> ; error of allowing your users to
>>> register at
>>> ; the same address as a SIP provider.
>>>
>>>
>>>
>>> On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
 [servera]
 type=friend
 username=servera
 secret=servera
 host=10.30.2.5
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
>

[asterisk-users] Asterisk 11, SIP. OK to BYE goes to wrong ip/port combination

2013-07-02 Thread Alex Zarubin 
Hi all,

I've read several discussions about asterisk adding 'received' parameter to the 
top Via header.

In our case asterisk (release 11.4) gets BYE from sip proxy (with BYE top via 
header containing proxy ip address and port) but added 'received' parameter 
contains ip address from a 2nd Via (or from "From') and OK gets lost.

I'm just trying to adjust sip configuration that used to work for simple call 
scenarios (in 1.4, for example) for Asterisk 11.
Your input is appreciated.

Thank you.

Alex Zarubin


In sip.conf

[general]
nat = no
outboundproxy=PROXYipaddress:PROXYport

[CARRIER]
type=peer
host=CARRIERipaddress
port=CARRIERport
canreinvite=no

Outbound call from asterisk is established normally via outbound proxy. BYE 
coming from the CARRIER

<--- SIP read from UDP:PROXYipaddress:PROXYport --->
BYE sip:XYZ@ASTERISKipaddress:5060 SIP/2.0
Via: SIP/2.0/UDP PROXYipaddress:PROXYport;branch=z9hG4bK-whatever-rsMNRDPbOY.0
Via: SIP/2.0/UDP 
CARRIERipaddress:CARRIERport;branch=z9hG4bK+17d247e1b781cd07f8d5339588fb32091+127.0.0.1+1
From: 
;tag=127.0.0.1alUtKGp-07233+1+9362fc7+68f1babf
To: ;tag=as1ca814af
Call-ID: 146b538429a2380a68bb374543d40c6d@OURipaddress:5060
CSeq: 1035426164 BYE
Max-Forwards: 69
User-Agent: Alcatel-Lucent 5060 MGC-8 8.3.0.6.SP1.2
Content-Length: 0
Supported: replaces, 100rel

Asterisk adds received=CARRIERipaddress (taken either from 2nd Via or from 
'From') and sends OK to CARRIERipaddress:PROXYport.
This OK goes nowhere, carrier re-sends BYE several times...

<--- Transmitting (no NAT) to CARRIERipaddress:PROXYport --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
PROXYipaddress:PROXYport;branch=z9hG4bK-whatever-rsMNRDPbOY.0;received= 
CARRIERipaddress
Via: SIP/2.0/UDP 
CARRIERipaddress:CARRIERport;branch=z9hG4bK+17d247e1b781cd07f8d5339588fb32091+127.0.0.1+1
From: 
;tag=127.0.0.1alUtKGp-07233+1+9362fc7+68f1babf
To: ;tag=as1ca814af
Call-ID: 146b538429a2380a68bb374543d40c6d@OURipaddress:5060
CSeq: 1035426164 BYE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Length: 0


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Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Asghar Mohammad 
make a call and post cli log


On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> still the peer shows unreachable let me restart and give a try...
>
>
> On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad wrote:
>
>> *1st Location*
>> [manila]
>> type=peer
>> username=indman01
>> secret=indman01
>> host=10.30.2.5 <-- ip of 2nd location
>> port=5060
>> context=Manila
>> insecure=port,invite
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> directmedia=no
>> qualify=yes
>> disallow=all
>> allow=g729
>> allow=ulaw
>>
>> 1st location dialplan
>> exten => _2XXX,1,Dial(SIP/manila/${EXTEN} 
>> )
>> exten => _2XXX,n,Hangup
>>
>> *2nd Location*
>> [india]
>> type=friend
>> username=manind01
>> secret=manind01
>> host=dynamic
>> port=5060
>> context=10.20.111.48 <- ip of 1st location
>>  insecure=port,invite
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> directmedia=no
>> qualify=yes
>> nat=force_rport,comedia
>> disallow=all
>> allow=g729
>> allow=ulaw
>> allow=alaw
>>
>> 2st location dialplan
>> exten => _2XXX,1,Dial(SIP/india/${EXTEN} )
>> exten => _2XXX,n,Hangup
>>
>> then you should handle the call when it arrive in any server
>> let me know if it work.
>>
>>
>> On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> I tried creating two trunks with following,
>>> *1st Location*
>>> [10.30.2.5]
>>> type=friend
>>> username=indman01
>>> secret=indman01
>>> host=dynamic
>>> port=5060
>>> context=Manila
>>> insecure=port,invite
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> directmedia=no
>>> qualify=yes
>>> disallow=all
>>> allow=g729
>>> allow=ulaw
>>>
>>> *2nd Location*
>>> [10.20.111.48]
>>> type=friend
>>> username=manind01
>>> secret=manind01
>>> host=dynamic
>>> port=5060
>>> context=india
>>> insecure=port,invite
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> directmedia=no
>>> qualify=yes
>>> nat=force_rport,comedia
>>> disallow=all
>>> allow=g729
>>> allow=ulaw
>>> allow=alaw
>>>
>>> My dialplan is like this
>>> exten => _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN}
>>> )
>>> exten => _2XXX,n,Hangup
>>>
>>> And the output I get is
>>>  Executing [2001@Test:1] Dial("SIP/3081-27d2", "SIP/10.30.2.5/2001")
>>> in new stack
>>> [Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
>>> dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
>>> Subscriber absent)
>>>   == Everyone is busy/congested at this time (1:0/0/1)
>>> -- Executing [2001@Test:2] Hangup("SIP/3081-27d2", "") in new
>>> stack
>>>   == Spawn extension (Test, 2001, 2) exited non-zero on
>>> 'SIP/3081-27d2'
>>>
>>> Actually the trunk which i mentioned in my first email, it was
>>> working... and from today it is not
>>>
>>> Still breaking... what could be the reason... !
>>>
>>>
>>>
>>> On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad wrote:
>>>
 yes you can. just create trunks on both side with static ip and in dial
 use trunk name.
 exten => _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =>
 _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
 make a call from a to b and one from b to and post cli log here or
 upload anyware else.


 On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N <
 gopalakrishnan...@gmail.com> wrote:

> can't we use without register command both way as peer to peer?
>
>
> On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad 
> wrote:
>
>> 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b
>> and 10.10.10.0 on a.
>> 2. use host=dynamic type=friend on  side A and host=ip type=peer on
>> side B.
>> 3. general section in sip.conf of side B register with server A.
>>
>> please see comments in sip.conf
>> ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
>> registering
>> ; as any IP address used for staticly
>> defined
>> ; hosts.  This helps avoid the
>> configuration
>> ; error of allowing your users to
>> register at
>> ; the same address as a SIP provider.
>>
>>
>>
>> On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> [servera]
>>> type=friend
>>> username=servera
>>> secret=servera
>>> host=10.30.2.5
>>> port=5060
>>> context=Manila
>>> insecure=port,invite
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> directmedia=no
>>> qualify=yes
>>> disallow=all
>>> allow=g729
>>> allow=ulaw
>>> allow=alaw
>>> deny=0.0.0.0/0.0.0.0
>>> permit=10.30.2.5/255.255.255.0
>>>
>>> If i use host=dynamic, it wont communicate each other and will
>>> result to unmonitored
>>>
>>>
>>> and the IP segment is two different segme

Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Gopalakrishnan N 
still the peer shows unreachable let me restart and give a try...


On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad  wrote:

> *1st Location*
> [manila]
> type=peer
> username=indman01
> secret=indman01
> host=10.30.2.5 <-- ip of 2nd location
> port=5060
> context=Manila
> insecure=port,invite
> dtmfmode=rfc2833
> relaxdtmf=yes
> directmedia=no
> qualify=yes
> disallow=all
> allow=g729
> allow=ulaw
>
> 1st location dialplan
> exten => _2XXX,1,Dial(SIP/manila/${EXTEN} )
> exten => _2XXX,n,Hangup
>
> *2nd Location*
> [india]
> type=friend
> username=manind01
> secret=manind01
> host=dynamic
> port=5060
> context=10.20.111.48 <- ip of 1st location
>  insecure=port,invite
> dtmfmode=rfc2833
> relaxdtmf=yes
> directmedia=no
> qualify=yes
> nat=force_rport,comedia
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
>
> 2st location dialplan
> exten => _2XXX,1,Dial(SIP/india/${EXTEN} )
> exten => _2XXX,n,Hangup
>
> then you should handle the call when it arrive in any server
> let me know if it work.
>
>
> On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> I tried creating two trunks with following,
>> *1st Location*
>> [10.30.2.5]
>> type=friend
>> username=indman01
>> secret=indman01
>> host=dynamic
>> port=5060
>> context=Manila
>> insecure=port,invite
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> directmedia=no
>> qualify=yes
>> disallow=all
>> allow=g729
>> allow=ulaw
>>
>> *2nd Location*
>> [10.20.111.48]
>> type=friend
>> username=manind01
>> secret=manind01
>> host=dynamic
>> port=5060
>> context=india
>> insecure=port,invite
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> directmedia=no
>> qualify=yes
>> nat=force_rport,comedia
>> disallow=all
>> allow=g729
>> allow=ulaw
>> allow=alaw
>>
>> My dialplan is like this
>> exten => _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN}
>> )
>> exten => _2XXX,n,Hangup
>>
>> And the output I get is
>>  Executing [2001@Test:1] Dial("SIP/3081-27d2", "SIP/10.30.2.5/2001")
>> in new stack
>> [Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
>> dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
>> Subscriber absent)
>>   == Everyone is busy/congested at this time (1:0/0/1)
>> -- Executing [2001@Test:2] Hangup("SIP/3081-27d2", "") in new
>> stack
>>   == Spawn extension (Test, 2001, 2) exited non-zero on
>> 'SIP/3081-27d2'
>>
>> Actually the trunk which i mentioned in my first email, it was working...
>> and from today it is not
>>
>> Still breaking... what could be the reason... !
>>
>>
>>
>> On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad wrote:
>>
>>> yes you can. just create trunks on both side with static ip and in dial
>>> use trunk name.
>>> exten => _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =>
>>> _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
>>> make a call from a to b and one from b to and post cli log here or
>>> upload anyware else.
>>>
>>>
>>> On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
 can't we use without register command both way as peer to peer?


 On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad wrote:

> 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b
> and 10.10.10.0 on a.
> 2. use host=dynamic type=friend on  side A and host=ip type=peer on
> side B.
> 3. general section in sip.conf of side B register with server A.
>
> please see comments in sip.conf
> ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
> registering
> ; as any IP address used for staticly
> defined
> ; hosts.  This helps avoid the
> configuration
> ; error of allowing your users to
> register at
> ; the same address as a SIP provider.
>
>
>
> On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> [servera]
>> type=friend
>> username=servera
>> secret=servera
>> host=10.30.2.5
>> port=5060
>> context=Manila
>> insecure=port,invite
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> directmedia=no
>> qualify=yes
>> disallow=all
>> allow=g729
>> allow=ulaw
>> allow=alaw
>> deny=0.0.0.0/0.0.0.0
>> permit=10.30.2.5/255.255.255.0
>>
>> If i use host=dynamic, it wont communicate each other and will result
>> to unmonitored
>>
>>
>> and the IP segment is two different segment. where am able to ping
>> each other.
>>
>>
>>
>> On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad 
>> wrote:
>>
>>> hi,
>>> paste server a trunk also, if you want register why you are not
>>> using host=dynamic?
>>> both servers are on 10.10.10.0 ? if no then check

Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Asghar Mohammad 
*1st Location*
[manila]
type=peer
username=indman01
secret=indman01
host=10.30.2.5 <-- ip of 2nd location
port=5060
context=Manila
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
disallow=all
allow=g729
allow=ulaw

1st location dialplan
exten => _2XXX,1,Dial(SIP/manila/${EXTEN} )
exten => _2XXX,n,Hangup

*2nd Location*
[india]
type=friend
username=manind01
secret=manind01
host=dynamic
port=5060
context=10.20.111.48 <- ip of 1st location
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
nat=force_rport,comedia
disallow=all
allow=g729
allow=ulaw
allow=alaw

2st location dialplan
exten => _2XXX,1,Dial(SIP/india/${EXTEN} )
exten => _2XXX,n,Hangup

then you should handle the call when it arrive in any server
let me know if it work.


On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> I tried creating two trunks with following,
> *1st Location*
> [10.30.2.5]
> type=friend
> username=indman01
> secret=indman01
> host=dynamic
> port=5060
> context=Manila
> insecure=port,invite
> dtmfmode=rfc2833
> relaxdtmf=yes
> directmedia=no
> qualify=yes
> disallow=all
> allow=g729
> allow=ulaw
>
> *2nd Location*
> [10.20.111.48]
> type=friend
> username=manind01
> secret=manind01
> host=dynamic
> port=5060
> context=india
> insecure=port,invite
> dtmfmode=rfc2833
> relaxdtmf=yes
> directmedia=no
> qualify=yes
> nat=force_rport,comedia
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
>
> My dialplan is like this
> exten => _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN}
> )
> exten => _2XXX,n,Hangup
>
> And the output I get is
>  Executing [2001@Test:1] Dial("SIP/3081-27d2", "SIP/10.30.2.5/2001")
> in new stack
> [Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
> dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
> Subscriber absent)
>   == Everyone is busy/congested at this time (1:0/0/1)
> -- Executing [2001@Test:2] Hangup("SIP/3081-27d2", "") in new
> stack
>   == Spawn extension (Test, 2001, 2) exited non-zero on 'SIP/3081-27d2'
>
> Actually the trunk which i mentioned in my first email, it was working...
> and from today it is not
>
> Still breaking... what could be the reason... !
>
>
>
> On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad wrote:
>
>> yes you can. just create trunks on both side with static ip and in dial
>> use trunk name.
>> exten => _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =>
>> _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
>> make a call from a to b and one from b to and post cli log here or upload
>> anyware else.
>>
>>
>> On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> can't we use without register command both way as peer to peer?
>>>
>>>
>>> On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad wrote:
>>>
 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b
 and 10.10.10.0 on a.
 2. use host=dynamic type=friend on  side A and host=ip type=peer on
 side B.
 3. general section in sip.conf of side B register with server A.

 please see comments in sip.conf
 ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
 registering
 ; as any IP address used for staticly
 defined
 ; hosts.  This helps avoid the
 configuration
 ; error of allowing your users to
 register at
 ; the same address as a SIP provider.



 On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N <
 gopalakrishnan...@gmail.com> wrote:

> [servera]
> type=friend
> username=servera
> secret=servera
> host=10.30.2.5
> port=5060
> context=Manila
> insecure=port,invite
> dtmfmode=rfc2833
> relaxdtmf=yes
> directmedia=no
> qualify=yes
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> deny=0.0.0.0/0.0.0.0
> permit=10.30.2.5/255.255.255.0
>
> If i use host=dynamic, it wont communicate each other and will result
> to unmonitored
>
>
> and the IP segment is two different segment. where am able to ping
> each other.
>
>
>
> On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad 
> wrote:
>
>> hi,
>> paste server a trunk also, if you want register why you are not using
>> host=dynamic?
>> both servers are on 10.10.10.0 ? if no then check your deny permit
>> seting.
>>
>>
>> On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Also tried one more scenario, particularly from one IP to other IP
>>> not registering.
>>>
>>> For example like 10.10.10.5 to 10.20.10.5
>>>
>>> If it is 10.10.10.5 to 10.30.2.5 -

Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Gopalakrishnan N 
I tried creating two trunks with following,
*1st Location*
[10.30.2.5]
type=friend
username=indman01
secret=indman01
host=dynamic
port=5060
context=Manila
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
disallow=all
allow=g729
allow=ulaw

*2nd Location*
[10.20.111.48]
type=friend
username=manind01
secret=manind01
host=dynamic
port=5060
context=india
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
nat=force_rport,comedia
disallow=all
allow=g729
allow=ulaw
allow=alaw

My dialplan is like this
exten => _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN})
exten => _2XXX,n,Hangup

And the output I get is
 Executing [2001@Test:1] Dial("SIP/3081-27d2", "SIP/10.30.2.5/2001") in
new stack
[Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [2001@Test:2] Hangup("SIP/3081-27d2", "") in new stack
  == Spawn extension (Test, 2001, 2) exited non-zero on 'SIP/3081-27d2'

Actually the trunk which i mentioned in my first email, it was working...
and from today it is not

Still breaking... what could be the reason... !



On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad  wrote:

> yes you can. just create trunks on both side with static ip and in dial
> use trunk name.
> exten => _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =>
> _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
> make a call from a to b and one from b to and post cli log here or upload
> anyware else.
>
>
> On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> can't we use without register command both way as peer to peer?
>>
>>
>> On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad wrote:
>>
>>> 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b
>>> and 10.10.10.0 on a.
>>> 2. use host=dynamic type=friend on  side A and host=ip type=peer on side
>>> B.
>>> 3. general section in sip.conf of side B register with server A.
>>>
>>> please see comments in sip.conf
>>> ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
>>> registering
>>> ; as any IP address used for staticly
>>> defined
>>> ; hosts.  This helps avoid the
>>> configuration
>>> ; error of allowing your users to
>>> register at
>>> ; the same address as a SIP provider.
>>>
>>>
>>>
>>> On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
 [servera]
 type=friend
 username=servera
 secret=servera
 host=10.30.2.5
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.30.2.5/255.255.255.0

 If i use host=dynamic, it wont communicate each other and will result
 to unmonitored


 and the IP segment is two different segment. where am able to ping each
 other.



 On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad wrote:

> hi,
> paste server a trunk also, if you want register why you are not using
> host=dynamic?
> both servers are on 10.10.10.0 ? if no then check your deny permit
> seting.
>
>
> On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Also tried one more scenario, particularly from one IP to other IP
>> not registering.
>>
>> For example like 10.10.10.5 to 10.20.10.5
>>
>> If it is 10.10.10.5 to 10.30.2.5 - working
>> If it is 10.30.2.5 to 10.20.10.4 works fine.
>>
>> really strange... I suspect some issue on the network side...
>>
>> Problem is there is no packet loss.. with mtr it is fine, tracepath
>> is fine, ping is fine... :(
>>
>>
>> On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Am using Asterisk 11.2 in one location and 11.1 in another location.
>>>
>>> when I trunk between two servers, the status is unreachable.
>>>
>>> But with different server with 11.2 and 11.2 it works fine.
>>>
>>> I tried both IAX and SIP.
>>>
>>> the trunk in sip.conf what i have is,
>>> [serverb]
>>> type=friend
>>> username=serverb
>>> secret=serverb
>>> host=10.10.10.5
>>> port=5060
>>> context=default
>>> insecure=port,invite
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> directmedia=no
>>> qualify=3000
>>> nat=force_rport,comedia
>>> disallow=all
>>> allow=g729
>>> allow=ulaw
>>> allow=alaw
>>> deny=0.0.0.0/0.0.0.0
>>> permit=10.10.10.5/255.255.255.0
>>>
>>> Is there any issue

[asterisk-users] Presence Management - use of hint

2013-07-02 Thread Eloi Bail 
Hi,

I am working on Presence management on a SIP client.

I have something working based on SUBSCRIBE / NOTIFY mechanism and Asterisk
hints.
I know that an other solution could be implemented using peer to peer
SUBSCRIBE / PUBLISH mechanism.

I would like to understand the advantage and drawback of each solution.
My main concern is the case of a complex VOIP environment such as Asterisk
server to server connection.


As example if we have :

[client A] <--> [Asterisk 1] <---> [Asterisk 2]
<--->[client B]


Using hints and SUBSCRIBE / NOTIFY would it be possible for client A to be
notified of client B presence modification ?


If client A can call client B, it would make sense to have SUBSCRIBE /
PUBLISH with peer to peer mechanism working. Am I right ?


Thanks for your advice,

Eloi
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Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Asghar Mohammad 
yes you can. just create trunks on both side with static ip and in dial use
trunk name.
exten => _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =>
_X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
make a call from a to b and one from b to and post cli log here or upload
anyware else.


On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> can't we use without register command both way as peer to peer?
>
>
> On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad wrote:
>
>> 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b and
>> 10.10.10.0 on a.
>> 2. use host=dynamic type=friend on  side A and host=ip type=peer on side
>> B.
>> 3. general section in sip.conf of side B register with server A.
>>
>> please see comments in sip.conf
>> ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
>> registering
>> ; as any IP address used for staticly
>> defined
>> ; hosts.  This helps avoid the
>> configuration
>> ; error of allowing your users to
>> register at
>> ; the same address as a SIP provider.
>>
>>
>>
>> On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> [servera]
>>> type=friend
>>> username=servera
>>> secret=servera
>>> host=10.30.2.5
>>> port=5060
>>> context=Manila
>>> insecure=port,invite
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> directmedia=no
>>> qualify=yes
>>> disallow=all
>>> allow=g729
>>> allow=ulaw
>>> allow=alaw
>>> deny=0.0.0.0/0.0.0.0
>>> permit=10.30.2.5/255.255.255.0
>>>
>>> If i use host=dynamic, it wont communicate each other and will result to
>>> unmonitored
>>>
>>>
>>> and the IP segment is two different segment. where am able to ping each
>>> other.
>>>
>>>
>>>
>>> On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad wrote:
>>>
 hi,
 paste server a trunk also, if you want register why you are not using
 host=dynamic?
 both servers are on 10.10.10.0 ? if no then check your deny permit
 seting.


 On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
 gopalakrishnan...@gmail.com> wrote:

> Also tried one more scenario, particularly from one IP to other IP not
> registering.
>
> For example like 10.10.10.5 to 10.20.10.5
>
> If it is 10.10.10.5 to 10.30.2.5 - working
> If it is 10.30.2.5 to 10.20.10.4 works fine.
>
> really strange... I suspect some issue on the network side...
>
> Problem is there is no packet loss.. with mtr it is fine, tracepath is
> fine, ping is fine... :(
>
>
> On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Am using Asterisk 11.2 in one location and 11.1 in another location.
>>
>> when I trunk between two servers, the status is unreachable.
>>
>> But with different server with 11.2 and 11.2 it works fine.
>>
>> I tried both IAX and SIP.
>>
>> the trunk in sip.conf what i have is,
>> [serverb]
>> type=friend
>> username=serverb
>> secret=serverb
>> host=10.10.10.5
>> port=5060
>> context=default
>> insecure=port,invite
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> directmedia=no
>> qualify=3000
>> nat=force_rport,comedia
>> disallow=all
>> allow=g729
>> allow=ulaw
>> allow=alaw
>> deny=0.0.0.0/0.0.0.0
>> permit=10.10.10.5/255.255.255.0
>>
>> Is there any issue with 11.1?
>>
>
>
> --
> _
> -- 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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 asterisk-users mailing list
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>>>
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>> --
>> _
>>

Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Gopalakrishnan N 
can't we use without register command both way as peer to peer?


On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad  wrote:

> 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b and
> 10.10.10.0 on a.
> 2. use host=dynamic type=friend on  side A and host=ip type=peer on side B.
> 3. general section in sip.conf of side B register with server A.
>
> please see comments in sip.conf
> ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
> registering
> ; as any IP address used for staticly
> defined
> ; hosts.  This helps avoid the
> configuration
> ; error of allowing your users to register
> at
> ; the same address as a SIP provider.
>
>
>
> On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> [servera]
>> type=friend
>> username=servera
>> secret=servera
>> host=10.30.2.5
>> port=5060
>> context=Manila
>> insecure=port,invite
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> directmedia=no
>> qualify=yes
>> disallow=all
>> allow=g729
>> allow=ulaw
>> allow=alaw
>> deny=0.0.0.0/0.0.0.0
>> permit=10.30.2.5/255.255.255.0
>>
>> If i use host=dynamic, it wont communicate each other and will result to
>> unmonitored
>>
>>
>> and the IP segment is two different segment. where am able to ping each
>> other.
>>
>>
>>
>> On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad wrote:
>>
>>> hi,
>>> paste server a trunk also, if you want register why you are not using
>>> host=dynamic?
>>> both servers are on 10.10.10.0 ? if no then check your deny permit
>>> seting.
>>>
>>>
>>> On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
 Also tried one more scenario, particularly from one IP to other IP not
 registering.

 For example like 10.10.10.5 to 10.20.10.5

 If it is 10.10.10.5 to 10.30.2.5 - working
 If it is 10.30.2.5 to 10.20.10.4 works fine.

 really strange... I suspect some issue on the network side...

 Problem is there is no packet loss.. with mtr it is fine, tracepath is
 fine, ping is fine... :(


 On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N <
 gopalakrishnan...@gmail.com> wrote:

> Am using Asterisk 11.2 in one location and 11.1 in another location.
>
> when I trunk between two servers, the status is unreachable.
>
> But with different server with 11.2 and 11.2 it works fine.
>
> I tried both IAX and SIP.
>
> the trunk in sip.conf what i have is,
> [serverb]
> type=friend
> username=serverb
> secret=serverb
> host=10.10.10.5
> port=5060
> context=default
> insecure=port,invite
> dtmfmode=rfc2833
> relaxdtmf=yes
> directmedia=no
> qualify=3000
> nat=force_rport,comedia
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> deny=0.0.0.0/0.0.0.0
> permit=10.10.10.5/255.255.255.0
>
> Is there any issue with 11.1?
>


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>>>
>>>
>>> --
>>> _
>>> -- 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
>>
>
>
> --
> _
> -- 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
>
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[asterisk-users] strange NAT issue?

2013-07-02 Thread Matt Hamilton 

We have a couple of cisco SPA phones and 3CX softphones behind a NAT firewall 
in a remote location. Firewall is connected to a bridged router which connects 
them to the public internet.

Router 5.6.7.8
Firewall 5.6.7.9(gateway 5.6.7.8)

Cisco SPA phone 192.168.1.4
Softphone  192.168.1.5


When these phones try to register, this is what we see on the Asterisk side:

The source IP of the softphones appear to be 5.6.7.9 (ip of the firewall) 
whereas the source IP of the hard phones are 5.6.7.8 (ip of the router) for 
some reason. What might cause the discrepancy? We would like to have the source 
to show up as 5.6.7.9 (IP of the remote firewall). 

Thanks,
Matt



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Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Asghar Mohammad 
1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b and
10.10.10.0 on a.
2. use host=dynamic type=friend on  side A and host=ip type=peer on side B.
3. general section in sip.conf of side B register with server A.

please see comments in sip.conf
;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
registering
; as any IP address used for staticly
defined
; hosts.  This helps avoid the configuration
; error of allowing your users to register
at
; the same address as a SIP provider.



On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> [servera]
> type=friend
> username=servera
> secret=servera
> host=10.30.2.5
> port=5060
> context=Manila
> insecure=port,invite
> dtmfmode=rfc2833
> relaxdtmf=yes
> directmedia=no
> qualify=yes
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> deny=0.0.0.0/0.0.0.0
> permit=10.30.2.5/255.255.255.0
>
> If i use host=dynamic, it wont communicate each other and will result to
> unmonitored
>
>
> and the IP segment is two different segment. where am able to ping each
> other.
>
>
>
> On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad wrote:
>
>> hi,
>> paste server a trunk also, if you want register why you are not using
>> host=dynamic?
>> both servers are on 10.10.10.0 ? if no then check your deny permit seting.
>>
>>
>> On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Also tried one more scenario, particularly from one IP to other IP not
>>> registering.
>>>
>>> For example like 10.10.10.5 to 10.20.10.5
>>>
>>> If it is 10.10.10.5 to 10.30.2.5 - working
>>> If it is 10.30.2.5 to 10.20.10.4 works fine.
>>>
>>> really strange... I suspect some issue on the network side...
>>>
>>> Problem is there is no packet loss.. with mtr it is fine, tracepath is
>>> fine, ping is fine... :(
>>>
>>>
>>> On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
 Am using Asterisk 11.2 in one location and 11.1 in another location.

 when I trunk between two servers, the status is unreachable.

 But with different server with 11.2 and 11.2 it works fine.

 I tried both IAX and SIP.

 the trunk in sip.conf what i have is,
 [serverb]
 type=friend
 username=serverb
 secret=serverb
 host=10.10.10.5
 port=5060
 context=default
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=3000
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.10.10.5/255.255.255.0

 Is there any issue with 11.1?

>>>
>>>
>>> --
>>> _
>>> -- 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
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
> asterisk-users mailing list
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>
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Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Gopalakrishnan N 
[servera]
type=friend
username=servera
secret=servera
host=10.30.2.5
port=5060
context=Manila
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=10.30.2.5/255.255.255.0

If i use host=dynamic, it wont communicate each other and will result to
unmonitored


and the IP segment is two different segment. where am able to ping each
other.



On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad  wrote:

> hi,
> paste server a trunk also, if you want register why you are not using
> host=dynamic?
> both servers are on 10.10.10.0 ? if no then check your deny permit seting.
>
>
> On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Also tried one more scenario, particularly from one IP to other IP not
>> registering.
>>
>> For example like 10.10.10.5 to 10.20.10.5
>>
>> If it is 10.10.10.5 to 10.30.2.5 - working
>> If it is 10.30.2.5 to 10.20.10.4 works fine.
>>
>> really strange... I suspect some issue on the network side...
>>
>> Problem is there is no packet loss.. with mtr it is fine, tracepath is
>> fine, ping is fine... :(
>>
>>
>> On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Am using Asterisk 11.2 in one location and 11.1 in another location.
>>>
>>> when I trunk between two servers, the status is unreachable.
>>>
>>> But with different server with 11.2 and 11.2 it works fine.
>>>
>>> I tried both IAX and SIP.
>>>
>>> the trunk in sip.conf what i have is,
>>> [serverb]
>>> type=friend
>>> username=serverb
>>> secret=serverb
>>> host=10.10.10.5
>>> port=5060
>>> context=default
>>> insecure=port,invite
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> directmedia=no
>>> qualify=3000
>>> nat=force_rport,comedia
>>> disallow=all
>>> allow=g729
>>> allow=ulaw
>>> allow=alaw
>>> deny=0.0.0.0/0.0.0.0
>>> permit=10.10.10.5/255.255.255.0
>>>
>>> Is there any issue with 11.1?
>>>
>>
>>
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Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Asghar Mohammad 
hi,
paste server a trunk also, if you want register why you are not using
host=dynamic?
both servers are on 10.10.10.0 ? if no then check your deny permit seting.


On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Also tried one more scenario, particularly from one IP to other IP not
> registering.
>
> For example like 10.10.10.5 to 10.20.10.5
>
> If it is 10.10.10.5 to 10.30.2.5 - working
> If it is 10.30.2.5 to 10.20.10.4 works fine.
>
> really strange... I suspect some issue on the network side...
>
> Problem is there is no packet loss.. with mtr it is fine, tracepath is
> fine, ping is fine... :(
>
>
> On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Am using Asterisk 11.2 in one location and 11.1 in another location.
>>
>> when I trunk between two servers, the status is unreachable.
>>
>> But with different server with 11.2 and 11.2 it works fine.
>>
>> I tried both IAX and SIP.
>>
>> the trunk in sip.conf what i have is,
>> [serverb]
>> type=friend
>> username=serverb
>> secret=serverb
>> host=10.10.10.5
>> port=5060
>> context=default
>> insecure=port,invite
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> directmedia=no
>> qualify=3000
>> nat=force_rport,comedia
>> disallow=all
>> allow=g729
>> allow=ulaw
>> allow=alaw
>> deny=0.0.0.0/0.0.0.0
>> permit=10.10.10.5/255.255.255.0
>>
>> Is there any issue with 11.1?
>>
>
>
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Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Gopalakrishnan N 
Also tried one more scenario, particularly from one IP to other IP not
registering.

For example like 10.10.10.5 to 10.20.10.5

If it is 10.10.10.5 to 10.30.2.5 - working
If it is 10.30.2.5 to 10.20.10.4 works fine.

really strange... I suspect some issue on the network side...

Problem is there is no packet loss.. with mtr it is fine, tracepath is
fine, ping is fine... :(


On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Am using Asterisk 11.2 in one location and 11.1 in another location.
>
> when I trunk between two servers, the status is unreachable.
>
> But with different server with 11.2 and 11.2 it works fine.
>
> I tried both IAX and SIP.
>
> the trunk in sip.conf what i have is,
> [serverb]
> type=friend
> username=serverb
> secret=serverb
> host=10.10.10.5
> port=5060
> context=default
> insecure=port,invite
> dtmfmode=rfc2833
> relaxdtmf=yes
> directmedia=no
> qualify=3000
> nat=force_rport,comedia
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> deny=0.0.0.0/0.0.0.0
> permit=10.10.10.5/255.255.255.0
>
> Is there any issue with 11.1?
>
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Re: [asterisk-users] where are stored Ip address of static sip peers

2013-07-02 Thread Grégory ESNAUD 
Other indication : there's no database registry on the main !

theMain*CLI> database show
/dundi/secret : 
7onlF5yX8Rgbk+htlh0AKQ==;QRi3PP1veWV0tc8WXRGHWA==
/dundi/secretexpiry   : 1366370052
2 results found.


theBackup*CLI> database show
/SIP/Registry/Z601-P3 : 
172.16.128.174:5060:3600:Z601-P3:sip:Z601-P3@172.16.128.174:5060;transport=udp
/SIP/Registry/Z602-P1 : 
172.16.128.167:5060:3600:Z602-P1:sip:Z602-P1@172.16.128.167:5060;transport=udp
/SIP/Registry/Z602-P2 : 
172.16.128.116:5060:3600:Z602-P2:sip:Z602-P2@172.16.128.116:5060;transport=udp
/SIP/Registry/Z602-P3 : 
172.16.128.170:5060:3600:Z602-P3:sip:Z602-P3@172.16.128.170:5060;transport=udp
/SIP/Registry/Z604-P1 : 
172.16.128.175:5060:3600:Z604-P1:sip:Z604-P1@172.16.128.175:5060;transport=udp
/SIP/Registry/Z604-P2 : 
172.16.128.157:5060:3600:Z604-P2:sip:Z604-P2@172.16.128.157:5060;transport=udp
/SIP/Registry/Z605-P1 : 
172.16.128.158:5060:3600:Z605-P1:sip:Z605-P1@172.16.128.158:5060;transport=udp
/SIP/Registry/Z605-P2 : 
172.16.128.176:5060:3600:Z605-P2:sip:Z605-P2@172.16.128.176:5060;transport=udp
/SIP/Registry/Z605-P3 : 
172.16.128.190:5060:3600:Z605-P3:sip:Z605-P3@172.16.128.190:5060;transport=udp
/SIP/Registry/Z607-P1 : 
172.16.128.172:5060:3600:Z607-P1:sip:Z607-P1@172.16.128.172:5060;transport=udp
/SIP/Registry/Z608-P1 : 
172.16.128.117:5060:3600:Z608-P1:sip:Z608-P1@172.16.128.117:5060;transport=udp
/SIP/Registry/Z608-P2 : 
172.16.128.153:5060:3600:Z608-P2:sip:Z608-P2@172.16.128.153:5060;transport=udp
/SIP/Registry/Z608-P3 : 
172.16.128.165:5060:3600:Z608-P3:sip:Z608-P3@172.16.128.165:5060;transport=udp
/SIP/Registry/Z608-P4 : 
172.16.128.161:5060:3600:Z608-P4:sip:Z608-P4@172.16.128.161:5060;transport=udp
/SIP/Registry/Z609-P1 : 
172.16.128.156:5060:3600:Z609-P1:sip:Z609-P1@172.16.128.156:5060;transport=udp
/SIP/Registry/Z609-P3 : 
172.16.128.171:5060:3600:Z609-P3:sip:Z609-P3@172.16.128.171:5060;transport=udp
/SIP/Registry/Z609-P4 : 
172.16.128.162:5060:3600:Z609-P4:sip:Z609-P4@172.16.128.162:5060;transport=udp
/dundi/secret : 
83BybhKWtGO4bqwNWdv+Tg==;mUsQNpnbfxrUfNQh8YDgdQ==
/dundi/secretexpiry   : 1372764840

So... why the database sip/registry is populated on the backup (that seems the 
normal behavior) and not on the main server?



De : asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Grégory ESNAUD
Envoyé : mardi 2 juillet 2013 10:04
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] where are stored Ip address of static sip peers

Hello dear list,

I came to you about a strange behavior on one of our server.

The architecture is built around 2 servers: a main and a backup. The clients 
(Cisco's Hardphone and counterpath 's softphone Bria) get the Asterisk ip 
addresses trough dns-srv mechanism (_sip.tcp).
The main Asterisk is located on our main site, and the backup is located on a 
remote site, so they are seperated through an operated WAN (Orange Business 
Services).
The same WAN is used to connect all of our sites (4), where clients are 
dispatched (hardphone and softphone).

The backup config files are a rsync copy of the main config files.
The softphone clients are realtime peers through an ldap setup.
The hardphone are static, configured through sip.conf.

The "strange" behavior is the following:
On the main server, we've got this (only showing the hardphone here):
theMain*CLI> sip show peers
Name/username HostDyn 
Forcerport ACL Port Status  Description  Realtime
Z601-P2/Z601-P2   (Unspecified)D
 0Unmonitored
Z601-P3/Z601-P3   (Unspecified)D
 0Unmonitored
Z602-P1/Z602-P1   (Unspecified)D
 0Unmonitored
Z602-P2/Z602-P2   (Unspecified)D
 0Unmonitored
Z602-P3/Z602-P3   (Unspecified)D
 0Unmonitored
Z604-P1/Z604-P1   (Unspecified)D
 0Unmonitored
Z604-P2/Z604-P2   (Unspecified)D
 0Unmonitored
Z605-P1/Z605-P1   (Unspecified)D

Re: [asterisk-users] Converting from FXO to SIP?

2013-07-02 Thread Mike Diehl 
Thank you!

Mike.

On Tue, Jul 2, 2013 at 1:37 PM, Administrator TOOTAI  wrote:
> Le 02/07/2013 21:06, Mike Diehl a écrit :
>>
>> [...]
>>
>> I was thinking that a TA with an FXO port might do the trick. But, I'm
>> not sure how to get the device to redirect an incoming call on the FXO
>> port to a sip destination. Is this something that gets done in the
>> device's dialplan?
>>
>> Does anyone have any insight into how to do this?
>>
>
> Hi Mike,
>
> you can do this without problem with any ATAs GW like Linksys SPA3102,
> SIPURA 3000, Tiger G102 aso. You just have to redirect FXO port to a SIP
> extension in the device and it's done. You can even redirect outgoing calls
> to this extension, for urgency calls like police, fire service, etc.
>
> Beronet (Berofix device) and others manufacturers have also gateways with
> FXO modules that you can use to fill your needs.
>
> --
> Daniel
>
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Re: [asterisk-users] Converting from FXO to SIP?

2013-07-02 Thread Administrator TOOTAI 

Le 02/07/2013 21:06, Mike Diehl a écrit :

[...]
I was thinking that a TA with an FXO port might do the trick. But, I'm
not sure how to get the device to redirect an incoming call on the FXO
port to a sip destination. Is this something that gets done in the
device's dialplan?

Does anyone have any insight into how to do this?



Hi Mike,

you can do this without problem with any ATAs GW like Linksys SPA3102, 
SIPURA 3000, Tiger G102 aso. You just have to redirect FXO port to a SIP 
extension in the device and it's done. You can even redirect outgoing 
calls to this extension, for urgency calls like police, fire service, etc.


Beronet (Berofix device) and others manufacturers have also gateways 
with FXO modules that you can use to fill your needs.


--
Daniel

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[asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Gopalakrishnan N 
Am using Asterisk 11.2 in one location and 11.1 in another location.

when I trunk between two servers, the status is unreachable.

But with different server with 11.2 and 11.2 it works fine.

I tried both IAX and SIP.

the trunk in sip.conf what i have is,
[serverb]
type=friend
username=serverb
secret=serverb
host=10.10.10.5
port=5060
context=default
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=3000
nat=force_rport,comedia
disallow=all
allow=g729
allow=ulaw
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=10.10.10.5/255.255.255.0

Is there any issue with 11.1?
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[asterisk-users] Converting from FXO to SIP?

2013-07-02 Thread Mike Diehl 
I have a customer who has an analog PBX that is able to be put in
"away mode" such that when an inbound call comes in, it rings their
cordless phone. This lets them leave the desk without risking missing
a call.

However, we'd like to find a black box that would act like the
cordless phone as far as the PBX was concerned, but instead of ringing
a handset, we want it to dial an extension on our SIP network. This
will allow us to handle the call much more flexibly than their PBX
can.

I was thinking that a TA with an FXO port might do the trick. But, I'm
not sure how to get the device to redirect an incoming call on the FXO
port to a sip destination. Is this something that gets done in the
device's dialplan?

Does anyone have any insight into how to do this?

TIA,

Mike Diehl

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[asterisk-users] Queue questions - Asterisk 11

2013-07-02 Thread Administrator TOOTAI 

Hi all,

I have to questions about queues. Member is a phone like SIP/myphone and 
only one member in the queue.


At first, DIALSTATUS doesn't return any status. How to now if a call in 
queue has been answered or if caller just hangup?


Second, how to deal with timeout, I have strange behaviors. If I put 
timeout=60 in queue.conf and I call the queue passing also 60 as timeout 
value, asterisk is returning after 5000ms the 4000 then 2000 then 2000 
aso. I can replace the 60sec value on both place, or 60 in queue conf 
and 10 when calling queue, I never have a stable behavior and more, not 
what I want.


Exemple: let say asterisk should try all 20 seconds to call the member 
for 8 seconds: how to configure this? What I found, is to put timeout=0 
in queue conf and passing 20 to queue, so caller stays in queue 20 
seconds before timeout. But asterisk rings 20 seconds :-(


I read documentation on voip-info.org, not very clear. timeout in 
queue.conf is only for calling agent, not members? If I put a timeout=8 
in queue conf, how to tell asterisk to retry each 20 seconds playing MOH 
to the caller?


Thanks for any hint

--
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Re: [asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"

2013-07-02 Thread Amit Patkar | ATPL 

Hi Matt,

As required, please find DEBUG trace for datetime function. I have used 
this function in Dialplan to capture DEBUG trace. I hope, this can help 
us in resolving the issue.


[Jul  2 15:54:44] DEBUG[2698] chan_sip.c: Checking device state for peer 
1001
[Jul  2 15:54:44] DEBUG[2698] devicestate.c: Changing state for SIP/1001 
- state 2 (In use)

[Jul  2 15:54:44] DEBUG[2698] devicestate.c: device 'SIP/1001' state '2'
[Jul  2 15:54:44] DEBUG[2737] pbx.c: Launching 'Answer'
[Jul  2 15:54:44] VERBOSE[2737] pbx.c: -- Executing [@avhan:1] 
Answer("SIP/1001-", "") in new stack
[Jul  2 15:54:44] DEBUG[2698] devicestate.c: No provider found, checking 
channel drivers for SIP - 1001
[Jul  2 15:54:44] DEBUG[2698] chan_sip.c: Checking device state for peer 
1001
[Jul  2 15:54:44] DEBUG[2698] devicestate.c: Changing state for SIP/1001 
- state 2 (In use)

[Jul  2 15:54:44] DEBUG[2698] devicestate.c: device 'SIP/1001' state '2'
[Jul  2 15:54:44] DEBUG[2737] chan_sip.c: SIP answering channel: 
SIP/1001-
[Jul  2 15:54:44] DEBUG[2737] res_rtp_asterisk.c: Setting the marker bit 
due to a source update
[Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Setting framing from config on 
incoming call
[Jul  2 15:54:44] DEBUG[2737] chan_sip.c: ** Our capability: 0x4 (ulaw) 
Video flag: True Text flag: True

[Jul  2 15:54:44] DEBUG[2737] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Jul  2 15:54:44] DEBUG[2737] chan_sip.c: -- Done with adding codecs to SDP
[Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Done building SDP. Settling 
with this capability: 0x4 (ulaw)
[Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Trying to put 'SIP/2.0 200' 
onto UDP socket destined for 192.168.2.18:7490
[Jul  2 15:54:44] DEBUG[2734] app_queue.c: Device 'SIP/1001' changed to 
state '2' (In use) but we don't care because they're not a member of any 
queue.
[Jul  2 15:54:44] DEBUG[2734] app_queue.c: Device 'SIP/1001' changed to 
state '2' (In use) but we don't care because they're not a member of any 
queue.
[Jul  2 15:54:44] DEBUG[2734] app_queue.c: Device 'SIP/1001' changed to 
state '2' (In use) but we don't care because they're not a member of any 
queue.
[Jul  2 15:54:44] DEBUG[2722] chan_sip.c: = Looking for  Call ID: 
YjNlMjU5YTJlMmQ5Njc3YjQ1MDgyMDg3ZjI1ZDViMmY. (Checking From) --From tag 
226b515a --To-tag as6e727cd7
[Jul  2 15:54:44] DEBUG[2722] chan_sip.c:  Received ACK (6) - 
Command in SIP ACK
[Jul  2 15:54:44] DEBUG[2722] chan_sip.c: Stopping retransmission on 
'YjNlMjU5YTJlMmQ5Njc3YjQ1MDgyMDg3ZjI1ZDViMmY.' of Response 2: Match Found

[Jul  2 15:54:44] DEBUG[2737] pbx.c: Launching 'DateTime'
[Jul  2 15:54:44] VERBOSE[2737] pbx.c: -- Executing [@avhan:2] 
DateTime("SIP/1001-", "136512,,YBd") in new stack
[Jul  2 15:54:44] DEBUG[2737] app_playback.c: string 
 depth <0>
[Jul  2 15:54:44] DEBUG[2737] app_playback.c: try 
 in 

[Jul  2 15:54:44] DEBUG[2737] pbx.c: Launching 'Hangup'
[Jul  2 15:54:44] VERBOSE[2737] pbx.c: -- Executing [@avhan:3] 
Hangup("SIP/1001-", "") in new stack
[Jul  2 15:54:44] DEBUG[2737] pbx.c: Spawn extension (avhan,,3) 
exited non-zero on 'SIP/1001-'
[Jul  2 15:54:44] VERBOSE[2737] pbx.c:   == Spawn extension (avhan, 
, 3) exited non-zero on 'SIP/1001-'
[Jul  2 15:54:44] DEBUG[2737] channel.c: Soft-Hanging up channel 
'SIP/1001-'
[Jul  2 15:54:44] DEBUG[2737] channel.c: Hanging up channel 
'SIP/1001-'
[Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Hangup call SIP/1001-, 
SIP callid YjNlMjU5YTJlMmQ5Njc3YjQ1MDgyMDg3ZjI1ZDViMmY.
[Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Updating call counter for 
incoming call
[Jul  2 15:54:44] DEBUG[2698] devicestate.c: No provider found, checking 
channel drivers for SIP - 1001
[Jul  2 15:54:44] DEBUG[2698] chan_sip.c: Checking device state for peer 
1001
[Jul  2 15:54:44] DEBUG[2698] devicestate.c: Changing state for SIP/1001 
- state 1 (Not in use)

[Jul  2 15:54:44] DEBUG[2698] devicestate.c: device 'SIP/1001' state '1'
[Jul  2 15:54:44] DEBUG[2737] res_rtp_asterisk.c: Setting RTCP address 
on RTP instance '0x98ac7f0'
[Jul  2 15:54:44] DEBUG[2737] netsock2.c: Splitting '192.168.2.18:7490' 
into...
[Jul  2 15:54:44] DEBUG[2737] netsock2.c: ...host '192.168.2.18' and 
port '7490'.
[Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Trying to put 'BYE sip:100' 
onto UDP socket destined for 192.168.2.18:7490


Thanks & Regards,
Amit Patkar

On 7/2/2013 5:15 PM, Matthew Jordan wrote:


On Tue, Jul 2, 2013 at 2:40 AM, Amit Patkar | ATPL > wrote:


Hello Matthew

I have pasted logs of the manager commands for the following
execution of the AGI Command and the result. As can be seen the
execution of the command replies "200 success" immediately without
executing the command. The date time is not played. Asterisk Logs
and AGI logs do not have anything of any significance , since we
use the asterisk manager.

This happens when we use mode=new , in say.c

Re: [asterisk-users] Endpoint call forwarding

2013-07-02 Thread Matthew Jordan 
On Tue, Jul 2, 2013 at 2:51 AM, John T. Bittner  wrote:

>  Anyone having issues with endpoint call forwarding on asterisk 11?
>
> ** **
>
> Was working perfect with 10. Issues are not phone related have tried
> cisco, polycom and Xlite, all fail.
>
> ** **
>
> Backtrack to 10 and it works ok again.
>
> ** **
>
> Any help is appreciated.
>
> ** **
>
> Thanks
>
> ** **
>
> ** **
>
> John Bittner
>
> CTO
>
>
>
I'm not aware of any issues currently open against call forwarding.

What specific problems are you seeing? What are the phones sending to
Asterisk, and what is Asterisk responding with?

A pastebin of a log showing DEBUG and higher level messages when a call
forward attempt occurs would help.

Thanks

Matt

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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[asterisk-users] where are stored Ip address of static sip peers

2013-07-02 Thread Grégory ESNAUD 
Hello dear list,

I came to you about a strange behavior on one of our server.

The architecture is built around 2 servers: a main and a backup. The clients 
(Cisco's Hardphone and counterpath 's softphone Bria) get the Asterisk ip 
addresses trough dns-srv mechanism (_sip.tcp).
The main Asterisk is located on our main site, and the backup is located on a 
remote site, so they are seperated through an operated WAN (Orange Business 
Services).
The same WAN is used to connect all of our sites (4), where clients are 
dispatched (hardphone and softphone).

The backup config files are a rsync copy of the main config files.
The softphone clients are realtime peers through an ldap setup.
The hardphone are static, configured through sip.conf.

The "strange" behavior is the following:
On the main server, we've got this (only showing the hardphone here):
theMain*CLI> sip show peers
Name/username HostDyn 
Forcerport ACL Port Status  Description  Realtime
Z601-P2/Z601-P2   (Unspecified)D
 0Unmonitored
Z601-P3/Z601-P3   (Unspecified)D
 0Unmonitored
Z602-P1/Z602-P1   (Unspecified)D
 0Unmonitored
Z602-P2/Z602-P2   (Unspecified)D
 0Unmonitored
Z602-P3/Z602-P3   (Unspecified)D
 0Unmonitored
Z604-P1/Z604-P1   (Unspecified)D
 0Unmonitored
Z604-P2/Z604-P2   (Unspecified)D
 0Unmonitored
Z605-P1/Z605-P1   (Unspecified)D
 0Unmonitored
Z605-P2/Z605-P2   (Unspecified)D
 0Unmonitored
Z605-P3/Z605-P3   (Unspecified)D
 0Unmonitored
Z607-P1/Z607-P1   (Unspecified)D
 0Unmonitored
Z608-P1/Z608-P1   (Unspecified)D
 0Unmonitored
Z608-P2/Z608-P2   (Unspecified)D
 0Unmonitored
Z608-P3/Z608-P3   (Unspecified)D
 0Unmonitored
Z608-P4/Z608-P4   (Unspecified)D
 0Unmonitored
Z609-P1/Z609-P1   (Unspecified)D
 0Unmonitored
Z609-P3/Z609-P3   (Unspecified)D
 0Unmonitored
Z609-P4/Z609-P4   (Unspecified)D
 0Unmonitored
Z609-P5/Z609-P5   (Unspecified)D
 0Unmonitored

That's normal, because the clients have been unplugged for many days (about 1 
week).

On the backup we've got this:
theBackup*CLI> sip show peers
Name/username HostDyn 
Forcerport ACL Port Status  Description  Realtime
Z601-P2/Z601-P2   (Unspecified)D
 0Unmonitored
Z601-P3/Z601-P3   172.16.128.174   D
 5060 Unmonitored
Z602-P1/Z602-P1   172.16.128.167   D
 5060 Unmonitored
Z602-P2/Z602-P2   172.16.128.116   D
 5060 Unmonitored
Z602-P3/Z602-P3   172.16.128.170   D
 5060 Unmonitored
Z604-P1/Z604-P1   172.16.128.175   D
 5060 Unmonitored
Z604-P2/Z604-P2   172.16.128.157   D
 5060 Unmonitored
Z605-P1/Z605-P1   172.16.128.158   D
 5060 Unmonitored
Z605-P2/Z605-P2   172.16.128.176   D
 5060 Unmonitored
Z605-P3/Z605-P3   172.16.128.190   D
 5060 Unmonitored
Z607-P1/Z607-P1   172.16.128.172   D
 5060 Unmonitored
Z608-P1/Z608-P1   172.16.128.117   D
 5060 Unmonitored
Z608-P2/Z608-P2   172.16.128.153   D
 5060 Unmonitored
Z608-P3/Z608-P3   172.16.128.165   D
 5060 Unmonitored
Z608-P4/Z608-P4   172.16.128.161   D
 5060 Unmonitored
Z609-P1/Z609-P1   172.

[asterisk-users] Endpoint call forwarding

2013-07-02 Thread John T. Bittner 
Anyone having issues with endpoint call forwarding on asterisk 11?

Was working perfect with 10. Issues are not phone related have tried cisco, 
polycom and Xlite, all fail.

Backtrack to 10 and it works ok again.

Any help is appreciated.

Thanks


John Bittner
CTO
[cid:image003.png@01CE76D7.8AB33690]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

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