Hello friends,
I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can
access the caller Call ID (fbasename field in voipmonitor cdr table)
looking at the SIPCALLID variable in asterisk, but how can I access from
within asterisk the Call ID of the second leg of the call (the one
Can you please show the database entry for that peer then?
On Thu, 2012-07-26 at 23:20 +0530, virendra bhati wrote:
My sip.conf don't have any entry related to sip pees. I have
everything into database.
for more details please check below url, which have good example of
asterisk realtime
Thanks a lot !
I will try the suggested solutions :)
Cheers!
pepesz
On Thu, Jul 26, 2012 at 3:02 PM, pepesz pep...@gmail.com wrote:
Dear all,
I know the topic comes back like boomerang, but I did not find a nice
solution.
Does someone has/knows how to achieve call back on busy
Hello friends,
I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can
access the caller Call ID (fbasename field in voipmonitor cdr table)
looking at the SIPCALLID variable in asterisk, but how can I access from
within asterisk the Call ID of the second leg of the call (the one
Hiii,
I am testing MWI on my grandstream and bria.
Following is sip show peer 1001
* Name : 1001
Secret : Set
MD5Secret: Not set
Remote Secret: Not set
Context : EXT_1001
Subscr.Cont. : Not set
Language :
AMA flags: Unknown
Transfer mode: open
On 07/26/2012 10:33 PM, Roi Stork wrote:
I've posted my problem with ReceiveFax() a long time ago.
Majority of the incoming faxes still end up with a T2 timeout or
hangup (fax session hangup) errors.
Our Setup:
- we're using the Digium Free Fax module for Asterisk, all settings are default
-
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Friday, July 27, 2012 10:45 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] still got ReceiveFax() problem, how to
On 07/27/2012 09:53 AM, Eric Wieling wrote:
People seem to think that Asterisk won't disable the Echo Canceler when a fax
tone is detected. Why they think that is a total mystery to me,.
Asterisk doesn't do this, the echo canceller itself does (or DAHDI does,
in some cases). With modern
- Original Message -
From: Dmitry Melekhov d...@belkam.com
To: asterisk-users@lists.digium.com
Sent: Thursday, July 26, 2012 10:40:57 PM
Subject: Re: [asterisk-users] Video conferencing?
25.07.2012 22:24, Ken D'Ambrosio пишет:
Hi, all. I'm 99% sure that Asterisk technically
strange last night my serve had this issue but when next morning i check
with register 1000 sip account no issue has come
thanks for your reply
On Fri, Jul 27, 2012 at 1:30 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
Can you please show the database entry for that peer then?
On Thu,
- Original Message -
On 07/26/2012 03:32 PM, Danny Nicholas wrote:
Question 1 - I think asterisk only supports a limited set of
statuses
Asterisk does not *receive* presence updates from Polycom phones (or
really, non-Digium phones) at all. Instead, the presence (status)
updates
Another mystery for the list, hopefully someone has ideas on a fix... :)
I've got an Asterisk 1.8.12.0 system connected to a CAS T1 (ESF/B8ZS,
fractional 1-8). Outbound dialing works correctly, but while the call is in
progress, there is no 'ringing' heard by the end user. So, on a SIP phone
On Friday 27 Jul 2012, Tim Nelson wrote:
Another mystery for the list, hopefully someone has ideas on a fix...
:)
I've got an Asterisk 1.8.12.0 system connected to a CAS T1 (ESF/B8ZS,
fractional 1-8). Outbound dialing works correctly, but while the
call is in progress, there is no 'ringing'
I think its not inbound call its outgoing, and during call progress the
remote end events are not passing back to sip.
Mitul
On Jul 27, 2012 10:36 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org
wrote:
On Friday 27 Jul 2012, Tim Nelson wrote:
Another mystery for the list, hopefully someone
Ok, Im putting back echo cancellation, since there's no change at all in
the fax receiving success rate.
I'd like to focus back to the original topic.
My incoming faxes end up usually as either timeout or hangup error:
timeout - is this supposed to happen in an E1 line? Can the timeout
threshold
Verizon has put another good third party DSL supplier out of the DSL business.
Their mindset is to kill the competition and then kill DSL and copper
althogether in FIOS areas.
So am soon losing my static IP and I need to prepare for the change. I
currently have Asterisk running using, besides
16 matches
Mail list logo
