Re: [Asterisk-Users] Help with Audicodes MP-104

2006-06-19 Thread Mahilal Silva
Not quite sure. Audiocodes gives a dialtone when the number is called from PSTN. After few seconds I see the SIP invite to the Asterisk box. Asterisk responds with SIP 404 .Thanks,Lal
On 6/12/06, Erick Perez [EMAIL PROTECTED] wrote:
So is the problem with your audiocodes or with the asterisk system?if it is with the asterisk, what kind of calls are you trying route toyour box? SIP/IAX/other?On 6/12/06, Mahilal Silva 
[EMAIL PROTECTED] wrote: Hi All I have been able to get MP 104 FXO to make outbound calls with my asterisk box and polycom IP 500 phone. However I cannot get the incoming calls to hit the asterisk box.
 Any help will be appreciated. Thanks, Lal ___ --Bandwidth and Colocation provided by Easynews.com
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[Asterisk-Users] Help with Audicodes MP-104

2006-06-12 Thread Mahilal Silva
Hi All
I have been able to get MP 104 FXO to make outbound calls with my asterisk box and polycom IP 500 phone.
However I cannot get the incoming calls to hit the asterisk box.
Any help will be appreciated.

Thanks,
Lal
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Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call

2006-02-24 Thread Mahilal Silva
Mike,
Were you able to get this working?
Even after with a entry in the dialplan.xml does not work for me.

Thanks,
Ken
On 6/20/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED] wrote:
Andrew,I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide?When you say mapped, dou mean that it needs an explicit entry in the
dialplan.xml like: TEMPLATE MATCH=# Timeout=0 User=Phone/ !--Explicit # for Asterisk --Mike- Original Message -
From: Andrew Latham [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comSent: Thursday, June 16, 2005 2:53 PMSubject: Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get #towork during a call# and * are mapped later in the SIP(Default/MAC).cnf it has a section
in the manual if you want to see why.On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED]wrote: Gents, I've built an Asterisk system to replace our PBX at work and have Cisco
 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001]
 type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby 2001 host=dynamic nat=no
 canreinvite=no dtmfmode=rfc2833 incominglimit=2 [EMAIL PROTECTED] disallow=all allow=alaw allow=ulaw callgroup=2 pickupgroup=2
 and in the SIPDefault.cnf for the phones I have: # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default),
 avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3
 DTMF works for voicemail and for remote services over both analogue Zap channels and digital (ISDN) channels. Asterisk doesn't appear to be 'monitoring' the audio so I can't get to Asterisk
 features like Asterisk's transfer, parked calls and one-tuch-record... Am I missing something? Mike ___
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--sigAndrew Latham - AKA: LATHAMA (lay-th-ham-eh)WWW: http://lathama.comEmail: [EMAIL PROTECTED] - 
[EMAIL PROTECTED] - [EMAIL PROTECTED]If any of the above are down we have bigger problems than my email!/sig___
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