VoIP Mailing List Archives

[stevej456 at gmail.com]

Please post your sip.conf entry for your phone and also describe your
calling path. Are you having a problem with internal calls (e.g.: to
voicemailmain) on the same switch, or are you referring to calls to
PSTN destinations via pots/pri/sip/? Also, which versions of
Asterisk, Zaptel, linux, etc. are you using?

S.

On Jan 24, 2008 12:43 PM, Jarga Jallow <Jarga at 2mcctv.com> wrote:

[Guest]

My polycoms all have dtmfmode=rfc2833 and they work fine on both
asterisk's IVRs and external ones brought to me from the PSTN:

[120]
type=friend
context=internalaugmented
secret=a_secret
host=dynamic
*dtmfmode=rfc2833*

Moj
Jarga Jallow wrote:

[Jarga at 2mcctv.com]

Sip.conf : ; Note: If your SIP devices are behind a NAT and your
Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.

[general]

bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf

I am calling other external phones, I think they PSTN destinations.

Jarga Jallow
Technical Support Engineer
2985 S. Hwy. 360
Grand Praire, Texas 75052
Direct: 972-206-1212 ext# 29
Mobile: 214-669-9046
Fax: 972-999-4113
Toll Free: 1-877-801-5511 ext 34
Toll Free: 1-877-926-2288


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve
Johnson
Sent: Thursday, January 24, 2008 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help: dtmf mode

Please post your sip.conf entry for your phone and also describe your
calling path. Are you having a problem with internal calls (e.g.: to
voicemailmain) on the same switch, or are you referring to calls to
PSTN destinations via pots/pri/sip/? Also, which versions of
Asterisk, Zaptel, linux, etc. are you using?

S.

On Jan 24, 2008 12:43 PM, Jarga Jallow <Jarga at 2mcctv.com> wrote:

[Jarga at 2mcctv.com]

On the polycom manual it says for g729 use rfc1890. I did that but
sometimes it works sometimes it doesn't. Am not sure why.

Jarga Jallow
Technical Support Engineer
2985 S. Hwy. 360
Grand Praire, Texas 75052
Direct: 972-206-1212 ext# 29
Mobile: 214-669-9046
Fax: 972-999-4113
Toll Free: 1-877-801-5511 ext 34
Toll Free: 1-877-926-2288


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mojo with
Horan & Company, LLC
Sent: Thursday, January 24, 2008 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help: dtmf mode

My polycoms all have dtmfmode=rfc2833 and they work fine on both
asterisk's IVRs and external ones brought to me from the PSTN:

[120]
type=friend
context=internalaugmented
secret=a_secret
host=dynamic
*dtmfmode=rfc2833*

Moj
Jarga Jallow wrote:

[gopal.krishnan at peop...]

Hi Jarga,

What type of connection you are using is it VoIP or ISDN PRI, if it is
VoIP check your dtmfmode in sip.conf if it is PRI check zapata.conf

On Jan 25, 2008 12:13 AM, Jarga Jallow <Jarga at 2mcctv.com> wrote: