Hi, I encounter this problem recently after quite some months of my asterisk. I have a SIP trunk for dial in and out. When dial-in, it matches the callerid number and decides. If matched, it will either go into an a very brief IVR. The IVR allows caller to dial internal extension. All along it is working well both from outside call and internal users. Now for unknown reason, it fails with a timeout and hangup. It is the only message I can see at the console. But internal user can do this without any problem. Appreciate if someone can help. CK -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110216/394fe655/attachment.htm>
Check if dtmfmode is properly set on SIP trunk ask with your carrier which dmtfmode they support. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, February 16, 2011 5:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DTMF not detected, time out Hi, I encounter this problem recently after quite some months of my asterisk. I have a SIP trunk for dial in and out. When dial-in, it matches the callerid number and decides. If matched, it will either go into an a very brief IVR. The IVR allows caller to dial internal extension. All along it is working well both from outside call and internal users. Now for unknown reason, it fails with a timeout and hangup. It is the only message I can see at the console. But internal user can do this without any problem. Appreciate if someone can help. CK -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110216/2cd2e3a6/attachment.htm>
In the past it was set as auto and worked. I change to RFC2833 but did not work. How can I check further? On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif <faisal at vopium.com> wrote:> Check if dtmfmode is properly set on SIP trunk ask with your carrier which > dmtfmode they support. > > > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *asterisk asterisk > *Sent:* Wednesday, February 16, 2011 5:39 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] DTMF not detected, time out > > > > Hi, > > I encounter this problem recently after quite some months of my asterisk. > > I have a SIP trunk for dial in and out. > When dial-in, it matches the callerid number and decides. If matched, it > will either go into an a very brief IVR. The IVR allows caller to dial > internal extension. > All along it is working well both from outside call and internal users. > Now for unknown reason, it fails with a timeout and hangup. It is the only > message I can see at the console. > But internal user can do this without any problem. > > Appreciate if someone can help. > > CK > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110216/bfd580e6/attachment.htm>
Ask with you SIP carrier which dtmfmode they are using on their end and use
same on asterisk side.
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk
asterisk
Sent: Wednesday, February 16, 2011 8:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF not detected, time out
In the past it was set as auto and worked. I change to RFC2833 but did not
work.
How can I check further?
On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif <faisal at vopium.com>
wrote:
Check if dtmfmode is properly set on SIP trunk ask with your carrier which
dmtfmode they support.
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk
asterisk
Sent: Wednesday, February 16, 2011 5:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DTMF not detected, time out
Hi,
I encounter this problem recently after quite some months of my asterisk.
I have a SIP trunk for dial in and out.
When dial-in, it matches the callerid number and decides. If matched, it
will either go into an a very brief IVR. The IVR allows caller to dial
internal extension.
All along it is working well both from outside call and internal users.
Now for unknown reason, it fails with a timeout and hangup. It is the only
message I can see at the console.
But internal user can do this without any problem.
Appreciate if someone can help.
CK
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20110216/c9a46e9b/attachment.htm>
You can also append add dtmf logging to cosole and see if dtmf is coming
from carrier.
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk
asterisk
Sent: Wednesday, February 16, 2011 8:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF not detected, time out
In the past it was set as auto and worked. I change to RFC2833 but did not
work.
How can I check further?
On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif <faisal at vopium.com>
wrote:
Check if dtmfmode is properly set on SIP trunk ask with your carrier which
dmtfmode they support.
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk
asterisk
Sent: Wednesday, February 16, 2011 5:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DTMF not detected, time out
Hi,
I encounter this problem recently after quite some months of my asterisk.
I have a SIP trunk for dial in and out.
When dial-in, it matches the callerid number and decides. If matched, it
will either go into an a very brief IVR. The IVR allows caller to dial
internal extension.
All along it is working well both from outside call and internal users.
Now for unknown reason, it fails with a timeout and hangup. It is the only
message I can see at the console.
But internal user can do this without any problem.
Appreciate if someone can help.
CK
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20110216/345d2b99/attachment.htm>
It is somehow back to normal. Nothing change. May be the sip provider problem. However, it lasts for quite a while. Thanks On Wed, Feb 16, 2011 at 12:04 PM, Faisal Hanif <faisal at vopium.com> wrote:> You can also append add dtmf logging to cosole and see if dtmf is coming > from carrier. > > > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *asterisk asterisk > *Sent:* Wednesday, February 16, 2011 8:58 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] DTMF not detected, time out > > > > In the past it was set as auto and worked. I change to RFC2833 but did not > work. > > How can I check further? > > > On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif <faisal at vopium.com> wrote: > > Check if dtmfmode is properly set on SIP trunk ask with your carrier which > dmtfmode they support. > > > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *asterisk asterisk > *Sent:* Wednesday, February 16, 2011 5:39 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] DTMF not detected, time out > > > > Hi, > > I encounter this problem recently after quite some months of my asterisk. > > I have a SIP trunk for dial in and out. > When dial-in, it matches the callerid number and decides. If matched, it > will either go into an a very brief IVR. The IVR allows caller to dial > internal extension. > All along it is working well both from outside call and internal users. > Now for unknown reason, it fails with a timeout and hangup. It is the only > message I can see at the console. > But internal user can do this without any problem. > > Appreciate if someone can help. > > CK > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110216/4d2f5c56/attachment.htm>
In your sip.conf, in trunk parameters use:
dtmfmode = INFO
Date: Wed, 16 Feb 2011 23:07:16 +0800
From: asterisk at ck-lee.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] DTMF not detected, time out
It is somehow back to normal. Nothing change. May be the sip provider problem.
However, it lasts for quite a while.
Thanks
On Wed, Feb 16, 2011 at 12:04 PM, Faisal Hanif <faisal at vopium.com>
wrote:
You can also append add dtmf logging to cosole and see if dtmf is coming from
carrier.
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of asterisk asterisk
Sent: Wednesday, February 16, 2011 8:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF not detected, time out
In the past it was set as auto and worked. I change to RFC2833 but did not
work.
How can I check further?
On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif <faisal at vopium.com>
wrote:Check if dtmfmode is properly set on SIP trunk ask with your carrier which
dmtfmode they support.
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of asterisk asterisk
Sent: Wednesday, February 16, 2011 5:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DTMF not detected, time out
Hi,
I encounter this problem recently after quite some months of my asterisk.
I have a SIP trunk for dial in and out.
When dial-in, it matches the callerid number and decides. If matched, it will
either go into an a very brief IVR. The IVR allows caller to dial internal
extension.
All along it is working well both from outside call and internal users.
Now for unknown reason, it fails with a timeout and hangup. It is the only
message I can see at the console.
But internal user can do this without any problem.
Appreciate if someone can help.
CK
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20110216/d6284fa3/attachment.htm>
some outside sip provider does not accept dtmf, if you have not this problem in your local, ask your outside carrier best On Wed, Feb 16, 2011 at 7:27 AM, asterisk asterisk <asterisk at ck-lee.com>wrote:> In the past it was set as auto and worked. I change to RFC2833 but did not > work. > > How can I check further? > > > > On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif <faisal at vopium.com> wrote: > >> Check if dtmfmode is properly set on SIP trunk ask with your carrier which >> dmtfmode they support. >> >> >> >> *From:* asterisk-users-bounces at lists.digium.com [mailto: >> asterisk-users-bounces at lists.digium.com] *On Behalf Of *asterisk asterisk >> *Sent:* Wednesday, February 16, 2011 5:39 AM >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* [asterisk-users] DTMF not detected, time out >> >> >> >> Hi, >> >> I encounter this problem recently after quite some months of my asterisk. >> >> I have a SIP trunk for dial in and out. >> When dial-in, it matches the callerid number and decides. If matched, it >> will either go into an a very brief IVR. The IVR allows caller to dial >> internal extension. >> All along it is working well both from outside call and internal users. >> Now for unknown reason, it fails with a timeout and hangup. It is the only >> message I can see at the console. >> But internal user can do this without any problem. >> >> Appreciate if someone can help. >> >> CK >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110217/1dd1ae34/attachment-0001.htm>