Hi,
On some SIP interconnects with devices like Cisco, Dialogic we get SIP
invite from different source port every time and asterisk rejects that
INVITE. Does anyone knows solution for this?
---
Kind Regards,
Deepika Nijhawan
VoIP Engineer
Oxygen8 Communications
T: +44(0
It just gives no matching peer error and doesn't pick their sip
configuration, so do not go to any context in extentions.conf.
VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from
IP:4604'
--
_
-- Bandwidth
It just gives no matching peer error and doesn't pick their sip
configuration, so do not go to any context in extentions.conf.
VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from
IP:4604'
--
_
-- Bandwidth
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Tuesday, June 15, 2010 9:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk reject SIP INTITE from different
source ports
Deepika Nijhawan wrote:
It just
It's working now after giving nat=yes, thanks.
Deepika
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Nijhawan
VoIP Engineer
Oxygen8 Communications
T: +44(0) 871 434 9151
+44(0) 121 620 9151
Email: deepika.nijha...@oxygen8.com
Skype: deepika-nijhawan
W: http://www.oxygen8.com/ www.oxygen8.com
This communication contains information which is confidential and may
Cdr status shows:
CDR logging: enabled
CDR mode: simple
CDR output unanswered calls: no
It is not showing 'CDR registered backend'
Thanks,
Deepika
From: Deepika Nijhawan [mailto:deepika.nijha...@oxygen8.com]
Sent: 18 June 2010 09:37
To: 'asterisk-users@lists.digium.com
Addons module is not installed. There is another pbx with just free pbx 2.7
installed and is showing cdrs on reports panel. So, wondering if I'm missing
some configuration on pbx which is not generating cdrs with free pbx 2.5
installed on that or is it because of the version.
Thanks,
Hi,
I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. Dahdi
status is not showing alarms but channels are not coming up. It is not
showing any channels when i run 'dahdi show channels'. Could anyone help
pls.
Thanks
Deepika
--
# service asterisk stop
# make install
# service asterisk start
Also, it is not showing speex translation on core show translation recalc
10.
Can anybody please tell if missing some step in this.
---
Kind Regards,
Deepika Nijhawan
VoIP Engineer
Oxygen8 Communications
Hi Chandrakant
I have checked and it shows func_speex module is enabled.
Where can I install speex-tools from ?
Asterisk version 1.6.2.10 and Centos 5.5 are installed.
---
Kind Regards,
Deepika Nijhawan
VoIP Engineer
Oxygen8 Communications
Hi,
Does anyone has an idea how to tell asterisk to use codec A for first 50
calls and then codec B for rest of the calls.
Thanks,
Deepika
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Ok. And how will we do for getting sip inbound calls from different ips and
sending them to dahdi.
Thanks,
D
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Hi,
Thanks. Actually can it be done on whole kit basis rather than for an
extension or peer. Like if there are lot of inbound sip interconnects on a
kit , how can we send first 50% simultaneous calls to dahdi with codec A and
after that with codec B.
Thanks,
D
--
Hi,
Group () and Group_Count () will need to be used on certain extension. What
if there are lot of clients on the kit with different routings some going to
dahdi and some to different sip interconnects, how can we do it on whole kit
basis. Or let me know if there is any other way to use
Hi,
I am trying to configure ipsec on asterisk. Have configured
/etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in
same folder.
Have run racoon. Still I can't receive calls.
Can anyone please tell if any extra step is needed.
Thanks
--
I am not getting anything in debug because call is not reaching us from
other end, it is inbound connection over ipsec.
Thanks.
From: Deepika Nijhawan [mailto:deepika.nijha...@oxygen8.com]
Sent: 08 September 2010 17:10
To: 'asterisk-users@lists.digium.com'
Subject: IPSec
Hi,
On freepbx (GUI), whatever reason number fails we always get 'all circuits
are busy' audio.
Does anybody know how to get far end audio when we dial wrong number or when
it's busy or unallocated number or failed with some other reason.
Thanks,
Deepika
--
Hi,
Asterisk is making a call to a peer. In 200 ok, peer is sending its
application server ip in contact field, so asterisk sends ACK to that IP.
RTP starts flowing between endpoints and peer plays an IVR and asks for
destination number. After entering destination number peer's application
Hi,
I am doing failover routing based on 2 dial commands. First route sends back
4xx response and I don't want it to try 2nd route when it is 4xx response.
Can we do failover routing based on SIP 5xx response only ?
Thanks
Deepika
--
Hi,
If I use dialstatus variable, it doesn't give exact reasons for failure like
for unallocated numbers it sends Congestion. Whereas, for unallocated number
I don't want to go to failover routing. But need to go to failover routing
for other congestion reasons.
So, is there any way to check
Ya, below is my routing:
Exten = 1234,1,Dial(SIP/abc)
Exten = 1234,n,Dial(SIP/xyz)
If 1234 is unallocated on SIP/abc it returns 1 in ${HANGUPCAUSE} variable.
For this I don't want it to try SIP/xyz.
But overall, if we get SIP 4xx reason then call should hangup like it sends
back 404 not found
Discussion
Cc: Deepika Nijhawan
Subject: Re: [asterisk-users] Failover Routing
On Tue, Mar 1, 2011 at 8:09 AM, Deepika Nijhawan
deepika.nijha...@oxygen8.com wrote:
Ya, below is my routing:
Exten = 1234,1,Dial(SIP/abc)
Exten = 1234,n,Dial(SIP/xyz)
If 1234 is unallocated on SIP/abc it returns 1
- Non-Commercial Discussion
Cc: Deepika Nijhawan
Subject: Re: [asterisk-users] Failover Routing
On Tue, Mar 1, 2011 at 8:09 AM, Deepika Nijhawan
deepika.nijha...@oxygen8.com wrote:
Ya, below is my routing:
Exten = 1234,1,Dial(SIP/abc)
Exten = 1234,n,Dial(SIP/xyz)
If 1234 is unallocated on SIP
Hangup cause gives ISDN cause codes.
Its easier with sip 4xx or 5xx as there are only 2 to check. So, looking to
get sip reason.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Thomas
Sent: 03 March 2011 02:28
To: Asterisk
TC400 and TCE400B digium cards that do codec translation,
1 How many of these cards can be installed on one server?
2 Can we combine it with the software codec aka for example if we had two
cards per server they could decompress 240 channels. However if we had
another say 100 calls on top of
26 matches
Mail list logo
