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Stefan Gofferje

2005-Jun-29 07:28 UTC

[Asterisk-Users] Play an announcement to the CALLING party

Hi folks,

how could I play an announcement to the calling party as soon, as the
called party picked up. I would like to deploy an asterisk in an
environment, where a premium rate support-number is offered to customers
which do not want to pay a monthly support contract. In Germany, you are
commited by law to announce the cost per minute of a premium rate number at
the beginning of the call. So, to avoid the employees forgetting it, an
automatic announcement should be played. Besides, same rules are applicable
for calls that may be recorded for quality assurance issues.
At least for premium rate calls, queues won't work as the customer would
strongly dislike hearing an announcement about the rate while waiting for
an agent.
The a() option of the dial app only works for CALLED parties and when
trying to use a macro with the m() option, the Playback also goes to the
called party.
Anyone any hints on that?

Regards,
Stefan


-- 
  (o_   Stefan Gofferje          | Linux Systems Specialist
  //\   Reg'd Linux User #247167 | Network Security Specialist
  V_/_  Heckler & Koch - the original point and click interface

Alexander Lopez

2005-Jun-29 08:23 UTC

head link

[Asterisk-Users] Play an announcement to the CALLING party

Why not play the message BEFORE you call the Dail application. This
would also give the caller a chance to terminiate the call by hanging up
BEFORE your techs even get the call..

Hint: use the playback application
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com 
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of 
> Stefan Gofferje
> Sent: Wednesday, June 29, 2005 10:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Play an announcement to the CALLING party
> 
> Hi folks,
> 
> how could I play an announcement to the calling party as 
> soon, as the called party picked up. I would like to deploy 
> an asterisk in an environment, where a premium rate 
> support-number is offered to customers which do not want to 
> pay a monthly support contract. In Germany, you are commited 
> by law to announce the cost per minute of a premium rate 
> number at the beginning of the call. So, to avoid the 
> employees forgetting it, an automatic announcement should be 
> played. Besides, same rules are applicable for calls that may 
> be recorded for quality assurance issues.
> At least for premium rate calls, queues won't work as the 
> customer would strongly dislike hearing an announcement about 
> the rate while waiting for an agent.
> The a() option of the dial app only works for CALLED parties 
> and when trying to use a macro with the m() option, the 
> Playback also goes to the called party.
> Anyone any hints on that?
> 
> Regards,
> Stefan
> 
> 
> -- 
>   (o_   Stefan Gofferje          | Linux Systems Specialist
>   //\   Reg'd Linux User #247167 | Network Security Specialist
>   V_/_  Heckler & Koch - the original point and click interface
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Alex Vishnev

2005-Jun-29 08:23 UTC

head link

[Asterisk-Users] Can't bridge between h323 and sip calls

Hello,

I am using asterisk CVS-head from 6/28. I am also using chan_oh323 that
comes with asterisk. I tried to place a call from h323 device into asterisk.
in extensions.conf, I routed the call to my sip phone. The sip phone was
already registered with asterisk. all the signaling looks ok to me. The sip
phone rings when h323 call hits the asterisk box. But then the call is
dropped. It appears that asterisk is trying to convert incoming g.729 codec
to ulaw and it can't. I was assumed that g.729 will pass-thru to the phone.
In fact, when an invite is sent bothg G729, G723 are codecs in SDP. However,
when SIP phone answers, it only replies with g723 on 200OK. I am still
unclear about that, but that's not really that important. I would like to
find out why I can't bridge these two legs. below is the trace from the
call. I am suspecting that a line below is the cause, but not sure why. Can
someone help???

Jun 29 10:59:46 WARNING[8862]: app_dial.c:1324 dial_exec_full: Had to drop
call because I couldn't make H323/ip$64.243.115.153:32971/11679 compatible
with SIP/debit-9f37

-----asterisk log------

    -- Executing Dial("H323/ip$64.243.115.153:32971/11679",
"SIP/debit|20|rt") in new stack
Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a
path from g729 to ulaw
Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a
path from g729 to ulaw
We're at 64.243.115.157 port 18192
Answering with capability 0x1 (g723)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with capability 0x100 (g729)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 13 lines
Reliably Transmitting (NAT) to 69.115.205.168:4152:
INVITE sip:debit@69.115.205.168:4146 SIP/2.0
Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport
From: "7323600296"
<sip:7323600296@64.243.115.157>;tag=as492d969f
To: <sip:debit@69.115.205.168:4146>
Contact: <sip:7323600296@64.243.115.157>
Call-ID: 0e4c2a05623aa0dd33497a316edf6671@64.243.115.157
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 29 Jun 2005 14:59:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 8862 8862 IN IP4 64.243.115.157
s=session
c=IN IP4 64.243.115.157
t=0 0
m=audio 18192 RTP/AVP 4 0 8 18 101
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called debit
Jun 29 10:59:41 WARNING[8862]: chan_h323.c:588 oh323_write: Asked to
transmit frame type 4, while native formats is 256 (read/write = 4/4)
Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a
path from g729 to slin
Jun 29 10:59:41 WARNING[8862]: indications.c:99 playtones_alloc: Unable to
set 'H323/ip$64.243.115.153:32971/11679' to signed linear format (write)
voip*CLI> 
<-- SIP read from 69.115.205.168:4152: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport
From: "7323600296"
<sip:7323600296@64.243.115.157>;tag=as492d969f
To: <sip:debit@192.168.15.175:4146>
Call-ID: 0e4c2a05623aa0dd33497a316edf6671@64.243.115.157
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.5.16
Warning: 399 69.115.205.168 "detected NAT type is symmetric NAT"
Content-Length: 0


--- (9 headers 0 lines)---
voip*CLI> 
<-- SIP read from 69.115.205.168:4152: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport
From: "7323600296"
<sip:7323600296@64.243.115.157>;tag=as492d969f
To: <sip:debit@192.168.15.175:4146>;tag=2cfc88182690d7d1
Call-ID: 0e4c2a05623aa0dd33497a316edf6671@64.243.115.157
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.5.16
Warning: 399 69.115.205.168 "detected NAT type is symmetric NAT"
Content-Length: 0


--- (9 headers 0 lines)---
    -- SIP/debit-9f37 is ringing
voip*CLI> 
<-- SIP read from 69.115.205.168:4152: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport
From: "7323600296"
<sip:7323600296@64.243.115.157>;tag=as492d969f
To: <sip:debit@192.168.15.175:4146>;tag=2cfc88182690d7d1
Call-ID: 0e4c2a05623aa0dd33497a316edf6671@64.243.115.157
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.5.16
Warning: 399 69.115.205.168 "detected NAT type is symmetric NAT"
Contact: <sip:debit@69.115.205.168:4146>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
Content-Length: 213

v=0
o=debit 0 8000 IN IP4 69.115.205.168
s=SIP Call
c=IN IP4 69.115.205.168
t=0 0
m=audio 4192 RTP/AVP 4 101
a=sendrecv
a=rtpmap:4 G723/8000
a=ptime:30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

--- (13 headers 11 lines)---
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 69.115.205.168:4192
Found description format G723
Found description format telephone-event
Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x1
(g723)/video=0x0 (nothing), combined - 0x1 (g723)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Jun 29 10:59:46 NOTICE[8862]: channel.c:1893 set_format: Unable to find a
path from g723 to ulaw
Jun 29 10:59:46 NOTICE[8862]: channel.c:1893 set_format: Unable to find a
path from g723 to ulaw
list_route: hop: <sip:debit@69.115.205.168:4146>
set_destination: Parsing <sip:debit@69.115.205.168:4146> for address/port
to
send to
set_destination: set destination to 69.115.205.168, port 4146
Transmitting (NAT) to 69.115.205.168:4152:
ACK sip:debit@69.115.205.168:4146 SIP/2.0
Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK21677c00;rport
From: "7323600296"
<sip:7323600296@64.243.115.157>;tag=as492d969f
o: <sip:debit@69.115.205.168:4146>;tag=2cfc88182690d7d1
Contact: <sip:7323600296@64.243.115.157>
Call-ID: 0e4c2a05623aa0dd33497a316edf6671@64.243.115.157
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
    -- SIP/debit-9f37 answered H323/ip$64.243.115.153:32971/11679
Jun 29 10:59:46 WARNING[8862]: channel.c:2317 ast_channel_make_compatible:
No path to translate from H323/ip$64.243.115.153:32971/11679(256) to
SIP/debit-9f37(1)
Jun 29 10:59:46 WARNING[8862]: app_dial.c:1324 dial_exec_full: Had to drop
call because I couldn't make H323/ip$64.243.115.153:32971/11679 compatible
with SIP/debit-9f37
set_destination: Parsing <sip:debit@69.115.205.168:4146> for address/port
to
send to
set_destination: set destination to 69.115.205.168, port 4146
Reliably Transmitting (NAT) to 69.115.205.168:4152:
BYE sip:debit@69.115.205.168:4146 SIP/2.0
Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK079d2b3d;rport
From: "7323600296"
<sip:7323600296@64.243.115.157>;tag=as492d969f
To: <sip:debit@69.115.205.168:4146>;tag=2cfc88182690d7d1
Contact: <sip:7323600296@64.243.115.157>
Call-ID: 0e4c2a05623aa0dd33497a316edf6671@64.243.115.157
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0


---
  == Spawn extension (default, 19087773456, 1) exited non-zero on
'H323/ip$64.243.115.153:32971/11679'
voip*CLI> 
<-- SIP read from 69.115.205.168:4152: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK079d2b3d;rport
From: "7323600296"
<sip:7323600296@64.243.115.157>;tag=as492d969f
To: <sip:debit@192.168.15.175:4146>;tag=2cfc88182690d7d1
Call-ID: 0e4c2a05623aa0dd33497a316edf6671@64.243.115.157
CSeq: 103 BYE
User-Agent: Grandstream BT100 1.0.5.16
Warning: 399 69.115.205.168 "detected NAT type is symmetric NAT"
Contact: <sip:debit@69.115.205.168:4146>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0


--- (12 headers 0 lines)---
Destroying call '0e4c2a05623aa0dd33497a316edf6671@64.243.115.157'
voip*CLI> sip no debug
SIP Debugging Disabled
voip*CLI>

Alex

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Stefan
Gofferje
Sent: Wednesday, June 29, 2005 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Play an announcement to the CALLING party

Hi folks,

how could I play an announcement to the calling party as soon, as the
called party picked up. I would like to deploy an asterisk in an
environment, where a premium rate support-number is offered to customers
which do not want to pay a monthly support contract. In Germany, you are
commited by law to announce the cost per minute of a premium rate number at
the beginning of the call. So, to avoid the employees forgetting it, an
automatic announcement should be played. Besides, same rules are applicable
for calls that may be recorded for quality assurance issues.
At least for premium rate calls, queues won't work as the customer would
strongly dislike hearing an announcement about the rate while waiting for
an agent.
The a() option of the dial app only works for CALLED parties and when
trying to use a macro with the m() option, the Playback also goes to the
called party.
Anyone any hints on that?

Regards,
Stefan


-- 
  (o_   Stefan Gofferje          | Linux Systems Specialist
  //\   Reg'd Linux User #247167 | Network Security Specialist
  V_/_  Heckler & Koch - the original point and click interface

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