Daniel - Asterisk
2009-Dec-02 17:32 UTC
[asterisk-users] Help configuring Audiocodes MP-104 FXO
Hi list,
I'm trying to get ready the MP-104 FXO to use qith my box, but when I send
calls I hear only dial tone and after a few seconds I get busy signal.
I very appreciate your advices.
Command line results and SIPconfigurations follows:
*CLI>*
-- Executing [7991696900 at total:1] Playback("SIP/101-09dd8918",
"beep")
in new stack
-- <SIP/101-09dd8918> Playing 'beep' (language 'es')
-- Executing [7991696900 at total:4] Dial("SIP/101-09dd8918",
"SIP/201/991696900") in new stack
-- Called 201/991696900
-- SIP/201-09ddc890 answered SIP/101-09dd8918
*sip.conf*
[201]
secret = ****
callerid = Mobile_01 <201>
type = friend
host = dynamic
context = total
dtmfmode=rfc2833
qualify = yes
call-limit=5
disallow = all
allow = gsm
allow = ulaw
allow = alaw
allow = g729
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Please post your BOARD.INI file (configuration of AudioCodes).
Also, do you expect to do single-stage dialing (MP104 takes SIP invite
information and turns that into DTMF output), or two-stage dialing
(MP104 only answers and connects the audio/RTP path)?
John Balogh, Sr. Systems Engineer
PSU, ITS, TNS, Network Planning
sip:jdb at psu.edu
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Daniel -
Asterisk
Sent: Wednesday, December 02, 2009 12:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Help configuring Audiocodes MP-104 FXO
Hi list,
I'm trying to get ready the MP-104 FXO to use qith my box, but when I
send calls I hear only dial tone and after a few seconds I get busy
signal.
I very appreciate your advices.
Command line results and SIPconfigurations follows:
CLI>
-- Executing [7991696900 at total:1] Playback("SIP/101-09dd8918",
"beep") in new stack
-- <SIP/101-09dd8918> Playing 'beep' (language 'es')
-- Executing [7991696900 at total:4] Dial("SIP/101-09dd8918",
"SIP/201/991696900") in new stack
-- Called 201/991696900
-- SIP/201-09ddc890 answered SIP/101-09dd8918
sip.conf
[201]
secret = ****
callerid = Mobile_01 <201>
type = friend
host = dynamic
context = total
dtmfmode=rfc2833
qualify = yes
call-limit=5
disallow = all
allow = gsm
allow = ulaw
allow = alaw
allow = g729
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> I want to do single-stage dialing. I've just realized I> have the two-stage running now (I get dial tone and then,> when i introduce the number, the call get through).Right. According to the SIP User's Manual LTRT-65405 MediaPack SIP User's Manual Ver 4.6.pdf page 67/294 " Enable Digit Delivery to Tel [EnableDigitDelivery] Disable [0] = Disabled (default). Enable [1] = Enable Digit Delivery feature for MediaPack/FXO & FXS. The digit delivery feature enables sending of DTMF digits to the gateway's port after the line is offhooked (FXS) or seized (FXO). For IP->Tel calls, after the line is offhooked / seized, the MediaPack plays the DTMF digits (of the called number) towards the phone line. [...] To use this feature with FXO gateways, configure the gateway to work in one stage dialing mode. " You probably need to set the above. The FXO parameter (from page 107/294): " Dialing Mode [IsTwoStageDial] One Stage [0] = One-stage dialing. Two Stage [1] = Two-stage dialing (default). Used for IP->FXO gateways calls. If two-stage dialing is enabled, then the FXO gateway seizes one of the PSTN/PBX lines without performing any dial, the remote user is connected over IP to PSTN/PBX, and all further signaling (dialing and Call Progress Tones) is performed directly with the PBX without the gateway's intervention. If one-stage dialing is enabled, then the FXO gateway seizes one of the available lines (according to Channel Select Mode parameter), and dials the destination phone number received in INVITE message. Use the 'Waiting For Dial Tone' parameter to specify whether the dialing should come after detection of dial tone, or immediately after seizing of the line. " So you probably need to clear that parameter (it is not configured in your .INI file now, so you need to add it, or change the web interface drop-down control). Let us know if this helps. JDB From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Daniel - Asterisk Sent: Wednesday, December 02, 2009 12:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help configuring Audiocodes MP-104 FXO Hi list, I'm trying to get ready the MP-104 FXO to use qith my box, but when I send calls I hear only dial tone and after a few seconds I get busy signal. I very appreciate your advices. Command line results and SIPconfigurations follows: CLI> -- Executing [7991696900 at total:1] Playback("SIP/101-09dd8918", "beep") in new stack -- <SIP/101-09dd8918> Playing 'beep' (language 'es') -- Executing [7991696900 at total:4] Dial("SIP/101-09dd8918", "SIP/201/991696900") in new stack -- Called 201/991696900 -- SIP/201-09ddc890 answered SIP/101-09dd8918 sip.conf [201] secret = **** callerid = Mobile_01 <201> type = friend host = dynamic context = total dtmfmode=rfc2833 qualify = yes call-limit=5 disallow = all allow = gsm allow = ulaw allow = alaw allow = g729 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091202/d67f3129/attachment.htm