[asterisk-users] FW: Call Xfer issue between DataCenter and User Site

2010-01-22 Thread Steven Davison
Sorry to bump this one...

Anyone have any other ideas on it?

Regards

Steven Davison
Net Technial Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Davison
Sent: 21 January 2010 08:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Xfer issue between DataCenter and User Site

Thanks for the responses on this one

David Gibbons: reinvite=no is set, as we need the asterisk box to maintain the 
audio for recording... (I believe even if we didn't have this option, 
MixMonitor would have the same effect anyway.)

Peder: the firewall is integrated into the router, and is a Zyxel 660H-D1... 
which hasn't caused NAT issues in the past, but it is something that we can 
switch out and see if a different make/model has the same problem.

In answer to your questions, the Data Center IP is the external address that 
has been 1 to 1 Nat'd to the internal address.

The phone site has no static Nat in place for Sip or RTP, so we are reliant on 
the routers ability to sort that out. There is a firewall on that router, which 
allows ALL traffic out, and also allows SIP and RTP in. 

Hope that clears up a few things! :)

Steven Davison - Network Engineer
t:   0845 0034567
f:   0845 0034543
w: www.ntsols.com


 
Net Technical Solutions | Suite 1 Wesley Chambers | Queens Road | Aldershot | 
Hampshire | GU11 3JD








   
 


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: 20 January 2010 18:24
To: asterisk-users@lists.digium.com
Cc: Alistair Mackenzie
Subject: Re: [asterisk-users] Call Xfer issue between DataCenter and User Site

Admittedly I didn't read your SIP debug (on the mobile), but do you have 
reinvite=no set for the extensions and SIP trunks (providers)?

This sounds on the surface like a classic case of the Mondays. Erm reinvites I 
mean.

snip
1. Incoming call from pstn/viop provider
2. Call is answered by a user
3. Call needs to be transferred
4. Xfer button is pushed, other user is called, answered, and they speak about 
the call
4b. The incoming call is held, listening to MoH
5. Xfer is pushed again,
6. SIP Debug Output
7. MoH stops,
8. Office user gets no audio
9. Incoming call is silent, and then call is dropped
10. Office user gets fed up of saying ‘hello??!?’ and hangs up.
/snip
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Re: [asterisk-users] Call Xfer issue between DataCenter and User Site

2010-01-21 Thread Steven Davison
Thanks for the responses on this one

David Gibbons: reinvite=no is set, as we need the asterisk box to maintain the 
audio for recording... (I believe even if we didn't have this option, 
MixMonitor would have the same effect anyway.)

Peder: the firewall is integrated into the router, and is a Zyxel 660H-D1... 
which hasn't caused NAT issues in the past, but it is something that we can 
switch out and see if a different make/model has the same problem.

In answer to your questions, the Data Center IP is the external address that 
has been 1 to 1 Nat'd to the internal address.

The phone site has no static Nat in place for Sip or RTP, so we are reliant on 
the routers ability to sort that out. There is a firewall on that router, which 
allows ALL traffic out, and also allows SIP and RTP in. 

Hope that clears up a few things! :)

Steven Davison - Network Engineer
t:   0845 0034567
f:   0845 0034543
w: www.ntsols.com


 
Net Technical Solutions | Suite 1 Wesley Chambers | Queens Road | Aldershot | 
Hampshire | GU11 3JD








   
 


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: 20 January 2010 18:24
To: asterisk-users@lists.digium.com
Cc: Alistair Mackenzie
Subject: Re: [asterisk-users] Call Xfer issue between DataCenter and User Site

Admittedly I didn't read your SIP debug (on the mobile), but do you have 
reinvite=no set for the extensions and SIP trunks (providers)?

This sounds on the surface like a classic case of the Mondays. Erm reinvites I 
mean.

snip
1. Incoming call from pstn/viop provider
2. Call is answered by a user
3. Call needs to be transferred
4. Xfer button is pushed, other user is called, answered, and they speak about 
the call
4b. The incoming call is held, listening to MoH
5. Xfer is pushed again,
6. SIP Debug Output
7. MoH stops,
8. Office user gets no audio
9. Incoming call is silent, and then call is dropped
10. Office user gets fed up of saying ‘hello??!?’ and hangs up.
/snip
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Re: [asterisk-users] DTMF reception during WaitForSilence

2010-01-21 Thread Steven Davison
You last question : why are DTMF tones not audible in the recording?

WE had issues with DTMF not recording, and found it was due to the handset only 
sending the DTMF in data, rather than inline, as a beep... that could be your 
reason :)

Steven Davison
Net Technial Solutions

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu
Sent: 21 January 2010 11:00
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF reception during WaitForSilence

Hello,

I wrote a little AGI-Script that implements an IVR (using asterisk 1.6).
The whole conversation is recorded and at some points the caller should 
tell some information.
I detect the silence (WaitForSilence) to go to the next step in the IVR. 
Until now everything is OK, but...
some information the user gives (or speaks) is numeric... some users 
have the habit, to enter numeric
information via the phonekeypad (ergo creating dtmf-tones) but I cant 
process DTMF-Input
during WaitForSilence.
How can I achive that both works simultaneously? I mean recording the 
spoken digits AND detecting
DTMF-Input AND detecting silence to know, when Input has finished... 
(I want to avoid that users
have to finish their input with the pound-key...) ?

Btw.: why are the DTMF-Tones, that a user enters, not hearable in the 
recording?

Thanks for your help and hints,
Yves

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Re: [asterisk-users] Caller hang up not detected

2010-01-21 Thread Steven Davison
Hi,

Couple of questions...

Are you allowing reinvites, and what happens if you change the dialplan to this?

exten = 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT)
exten = 1,n,Playback(vm-goodbye)
exten = 1,n,Hangup()

help this helps :)

Steven Davison
Net Technial Solutions

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hugolivude
Sent: 21 January 2010 13:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Caller hang up not detected

Hi,

I'm having trouble getting Dial to exit when the caller hangs up in Asterisk 
1.4.21.2.

I use a POTS line to call into the DiD given to me by VOIP service provider.  
When the call comes in, I have the VOIP provider send it to another POTS line.  
All this works fine however when the caller (me) hangs up, the Dial command 
does not exit.  The callee stays connected (and my billing continues!). Dial 
doesn't exit until the callee hangs up. Here's a snip from extensions.conf:

exten = 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT)
exten = 1,n,Playback(vm-goodbye)

Here's the CLI output (verbosity = 4):

-- Executing [...@trunk-0001:1] NoOp(SIP/77.57.127.163-09023590, ) in new 
stack
-- Executing [...@trunk-0001:2] Dial(SIP/77.57.127.163-09023590, 
SIP/14168724...@6135551212-sw1|120|gtT) in new stack
-- Called 14168724...@6135551212-sw1
-- SIP/6135551212-sw1-090275d0 is making progress passing it to 
SIP/77.57.127.163-09023590
-- SIP/6135551212-sw1-090275d0 answered SIP/77.57.127.163-09023590
*** I hang up here, but the call continues.  A while later the callee hangs up:
-- Executing [...@trunk-0001:3] Playback(SIP/77.57.127.163-09023590, 
vm-goodbye) in new stack
*** obviously I don't here this, just see it in the CLI

I'd be grateful for any troubleshooting tips that will help me get asterisk to 
quit the Dial command when the originator hangs up.

Thanks,
H
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[asterisk-users] Call Xfer issue between DataCenter and User Site

2010-01-20 Thread Steven Davison
Hi, 

I am running a Asterisk 1.6 box in our Data Centre, and have a number of users 
connecting to that box, as their PBX.

Calls in and out work fine, as does voicemail.

The PBX at the Data Centre has an External IP, Nat’d to it by the firewall, and 
the relevant ports are open.

The Office users have a dedicated internet connection for the phone lines, and 
calls are seen to traverse this correctly. The handsets are Linksys SPA922

The issue we are getting is in transferring calls, which happens like this :-

1. Incoming call from pstn/viop provider
2. Call is answered by a user
3. Call needs to be transferred
4. Xfer button is pushed, other user is called, answered, and they speak about 
the call
4b. The incoming call is held, listening to MoH
5. Xfer is pushed again,
6. SIP Debug Output
7. MoH stops, 
8. Office user gets no audio
9. Incoming call is silent, and then call is dropped
10. Office user gets fed up of saying ‘hello??!?’ and hangs up.

Here is the sip debug output...


[Jan 20 16:43:38] set_destination: Parsing sip:1...@xxx.xxx.xxx.xxx:10036 for 
address/port to send to
[Jan 20 16:43:38] set_destination: set destination to XXX.XXX.XXX.XXX, port 
10036
[Jan 20 16:43:38] Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:10016:
NOTIFY sip:1...@xxx.xxx.xxx.xxx:10036 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK48cff632;rport
Max-Forwards: 70
From: Steve (NetTech) sip:1...@yyy.yyy.yyy.yyy;tag=as4f7c4d0c
To: sip:1...@xxx.xxx.xxx.xxx:10036;tag=726be2fb618280d0i0
Contact: sip:1...@yyy.yyy.yyy.yyy
Call-ID: 718a30a4572984a918b88dc64df64...@yyy.yyy.yyy.yyy
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.6.1.1
Event: refer;id=102
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 49

SIP/2.0 481 Call leg/transaction does not exist

---
[Jan 20 16:43:38] -- Stopped music on hold on SIP/176-09bf9630
[Jan 20 16:43:38]
--- SIP read from UDP://XXX.XXX.XXX.XXX:10016 ---
SIP/2.0 200 OK
To: sip:1...@xxx.xxx.xxx.xxx:10036;tag=726be2fb618280d0i0
From: Steve (NetTech) sip:1...@yyy.yyy.yyy.yyy;tag=as4f7c4d0c
Call-ID: 718a30a4572984a918b88dc64df64...@yyy.yyy.yyy.yyy
CSeq: 103 NOTIFY
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK48cff632
Server: Linksys/SPA922-4.1.18
Content-Length: 0


-
YYY.YYY.YYY.YYY is the IP of the Datacenter
XXX.XXX.XXX.XXX is the IP of the Office

I have been going over and over the configs on the routers, sip.conf etc trying 
to work this out... we have also checked that the users are using the above 
sequence to transfer a call...

Thanks to anyone who may have ideas for this... ☺

Steven Davison - Network Engineer
t:   0845 0034567
f:   0845 0034543
w: www.ntsols.com


 
Net Technical Solutions | Suite 1 Wesley Chambers | Queens Road | Aldershot | 
Hampshire | GU11 3JD








   
 

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