Re: [asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code.

2017-03-09 Thread Jean Aunis 
This is not a SIP NOTIFY but a SIP MESSAGE (the first line in the logs 
is not related to the few next ones).


If you are using chan_sip, you have to activate out of call messages in 
sip.conf :


accept_outofcall_message=yes
outofcall_message_context=messages

Then in extensions.conf, define a context "messages" with the 
appropriate extensions (to stick to your example, it will be 
16162995607) and use the function MESSAGE to retrieve the SMS content.


Best regards

Jean Aunis


Le 10/03/2017 à 00:21, Bryant Zimmerman a écrit :

I am trying to send SMS from my grandstream GXV3240
Asterisk receives the message in a NOTIFY block.
How can I get asterisk to run dialplan code when receiving these 
Notify SMS Message Blocks.

I can then route them to my SMS provider.
Any ideas are appreciated. Below is debug of a message sent from the 
phone when received no dialplan code is triggered.
I am wounding if I need to modify some setting in sip.conf or the peer 
config.  Incomming SMS from my vendor works without issue and is 
transmitted to the phone.


<->
--- (10 headers 0 lines) ---
Really destroying SIP dialog 
'3927411c7fe967886df6c8d0410d4...@xxx.xxx.xxx.xxx:5060' 
 
Method: NOTIFY


<--- SIP read from UDP:XXX.XXX.XXX.XXX:57568 --->
MESSAGE sip:16162995...@vgw0005.granddial.net SIP/2.0
Via: SIP/2.0/UDP 192.168.201.104:20093;branch=z9hG4bK1738682353;rport
From: ;tag=1683585926
To: 
Call-ID: 1662412698-20093-...@bjc.bgi.cab.bae 


CSeq: 9430 MESSAGE
Contact: 
Max-Forwards: 70
User-Agent: Grandstream GXV3240 1.0.3.158
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, 
REFER, UPDATE, MESSAGE

Content-Type: text/plain; charset=UTF-8
Content-Length: 5

Test Message SMS
<->


Thanks

Bryant




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[asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code.

2017-03-09 Thread Bryant Zimmerman 
I am trying to send SMS from my grandstream GXV3240
 Asterisk receives the message in a NOTIFY block.
  
 How can I get asterisk to run dialplan code when receiving these Notify 
SMS Message Blocks.
 I can then route them to my SMS provider.
  
 Any ideas are appreciated. Below is debug of a message sent from the phone 
when received no dialplan code is triggered.
 I am wounding if I need to modify some setting in sip.conf or the peer 
config.  Incomming SMS from my vendor works without issue and is 
transmitted to the phone.
  

<->
--- (10 headers 0 lines) ---
Really destroying SIP dialog 
'3927411c7fe967886df6c8d0410d4...@xxx.xxx.xxx.xxx:5060' Method: NOTIFY  

<--- SIP read from UDP:XXX.XXX.XXX.XXX:57568 --->
MESSAGE sip:16162995...@vgw0005.granddial.net SIP/2.0
Via: SIP/2.0/UDP 192.168.201.104:20093;branch=z9hG4bK1738682353;rport
From: ;tag=1683585926
To: 
Call-ID: 1662412698-20093-...@bjc.bgi.cab.bae
CSeq: 9430 MESSAGE
Contact: 
Max-Forwards: 70
User-Agent: Grandstream GXV3240 1.0.3.158
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, 
UPDATE, MESSAGE
Content-Type: text/plain; charset=UTF-8
Content-Length: 5  

Test Message SMS
<-> 

Thanks

Bryant 

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[asterisk-users] Optimizing forwarded and redirected calls by avoiding signaling and media data redirection

2017-03-09 Thread Sree Harsha Totakura 
Hi!

I'm having a setup where my asterisk PBX connects to PSTN via a single
SIP trunk.  Now, when I transfer or redirect incoming calls from the SIP
trunk to another number which is routed through the SIP trunk, my
asterisk stays on the way; it just dials out the new destination number
the call is transferred/redirected to and connects the newly dialed
channel to the existing incoming channel.

Since these two channels are in the same SIP trunk, would it be possible
to tell the trunk SIP server to not involve my asterisk anymore, both
for signaling and media data?  Or is this inherently not possible via SIP?

Regards,
Sree

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[asterisk-users] asterisk Channel 'IAX2' unable to transfer

2017-03-09 Thread neu pat 
I've upgraded to to asterisk 11.25.1 (from 1.8).  My local asterisk is
showing it is registered with remote asterisk (same version),
But when I try to make a call I get:


iax2 show registry
Host  dnsmgr  UsernamePerceived Refresh  State
192.168.142.1:4569N   home_serve  192.168.142.7:4569
60  Registered

1 IAX2 registrations.
-- Accepted AUTHENTICATED TBD call from 10.0.0.108
-- Accepting DIAL from 10.0.0.108, formats = (ulaw)
-- Executing [4@internal:1] Dial("IAX2/iaxy-322-3730",
"IAX2/home_server:546987@192.168.141.1/4,30,rw") in new stack
-- Called IAX2/home_server:x@192.168.141.1/4
-- Call accepted by 192.168.141.1 (format ulaw)
-- Format for call is (ulaw)
-- IAX2/192.168.141.1:4569-83 answered IAX2/iaxy-322-3730
-- Channel 'IAX2/192.168.141.1:4569-83' unable to transfer
-- Channel 'IAX2/192.168.141.1:4569-83' unable to transfer
-- Hungup 'IAX2/192.168.141.1:4569-83'
  == Spawn extension (internal, 4, 1) exited non-zero on 'IAX2/iaxy-322-3730'
-- Hungup 'IAX2/iaxy-322-3730'

Is there something wrong with my dial plan:
exten => 4,1,Dial(IAX2/home_server:xx@${clinic_server}/${EXTEN},30,rw)

-- 
Regards,
Thelma

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[asterisk-users] tcpbind and source IP address

2017-03-09 Thread Kseniya Blashchuk 
Hi all!
I am running asterisk 13.1.0 on Ubuntu server 16.04. There are two IP
addresses from the same subnet set on one interface, and bindaddr is set to
the second on them in sip.conf and in iax.conf.
Incoming connections work as expected. However, for outgoing connections it
seems that asterisk tells the kernel to use the specific "bind" address
only in case of UDP usage (both SIP and IAX work like that). In case of
outgoing TCP connections (SIP TCP and TLS) the first IP address from the
interface is used.
In my understanding, normally 'bind' should not only tell on which address
to listen, but also which source address to request for outgoing
connections, but it works only for UDP connections for some reason.
Can anybody explain if it's a normal behavior?
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