>> check the system and make sure there really is no firewall like I said
> You were right.
Stick around on the list long enough and you'll realise the truth... he always
is ;-)
Pete
--
_
-- Bandwidth and Colocation Provide
You were right. I had non-default rtp ports open in iptables. Edited
rtp.conf et voila. Everything seems to be working.
Thanks so much for your patience and guidance!
Have a lovely eening.
--
_
-- Bandwidth and Colocation Provid
Chirag Desai wrote:
So I see:
EXTERNAL_SNOM_IP -> EXTERNAL_ASTERISK_IP (UDP, length 218, src:
60798, dst 11128)
EXTERNAL_ASTERISK_IP -> INTERNAL_SNOM_IP (UDP, length 218, src: 11128
dst 60478
So i see udp from the phone, but there's no audio.
If "rtp set debug on" shows no packets being
So I see:
EXTERNAL_SNOM_IP -> EXTERNAL_ASTERISK_IP (UDP, length 218, src: 60798,
dst 11128)
EXTERNAL_ASTERISK_IP -> INTERNAL_SNOM_IP (UDP, length 218, src: 11128 dst
60478
So i see udp from the phone, but there's no audio.
I do also see some packets ::
EXTERNAL_ASTERISK_IP -> EXTERNAL_SN
Chirag Desai wrote:
In the PCAP I can see asterisk sending UDP packets to my local IP
192.168.0.5
If you don't see anything arriving from the remote side and we've told
them the right IP address and ICE is not actually negotiated... then
that leans more towards something remote unless the
In the PCAP I can see asterisk sending UDP packets to my local IP
192.168.0.5
It's funny, when I switch to TCP on 5060 audio seems to work fine. The
moment I go to 5063 on TLS everything goes a bit awry. Any further input is
greatly appreciated.
--
Chirag Desai wrote:
I'm dialling from the snom and every few calls asterisk sends media to
the phones external IP and it works!
And then now and again it sends the media to the phones internal IP and
I hear nothing. I'm really at a loss.
In the non-working case check the IP address in the SDP,
I'm dialling from the snom and every few calls asterisk sends media to the
phones external IP and it works!
And then now and again it sends the media to the phones internal IP and I
hear nothing. I'm really at a loss.
--
_
-- Ban
Chirag Desai wrote:
Joshua Colp wrote:
Have you done a packet capture to see if the RTP from the remote device
is hitting the machine to narrow things down?
Nope. When I run with RTP encryption on it seems that rewrite_contact
does not work in PJSIP.
When I turn of
> Joshua Colp wrote:
>>
>> Have you done a packet capture to see if the RTP from the remote device
>> is hitting the machine to narrow things down?
>>
>>
>>
Nope. When I run with RTP encryption on it seems that rewrite_contact does
not work in PJSIP.
When I turn off RTP some calls get media, some
Chirag Desai wrote:
Joshua Colp wrote:
There should be nothing different, except for how you configure things.
What is the full PJSIP configuration? What is the environment where
Asterisk is running? Is ICE actually in use on the other side? What is
the full SIP trace?
Th
> Joshua Colp wrote:
>
> There should be nothing different, except for how you configure things.
> What is the full PJSIP configuration? What is the environment where
> Asterisk is running? Is ICE actually in use on the other side? What is
> the full SIP trace?
>
The full configuration is here:
Chirag Desai wrote:
I have two accounts on Asterisk 13. One uses chan_sip and the other pjsip.
In my snom 760 the setup for these two accounts is identical.
When I call echo test from the account using chan_sip audio comes
through fine.
When I call echo test from the account using pjsip there
I have two accounts on Asterisk 13. One uses chan_sip and the other pjsip.
In my snom 760 the setup for these two accounts is identical.
When I call echo test from the account using chan_sip audio comes through
fine.
When I call echo test from the account using pjsip there is no audio.
With rtp
Hello,
Sorry for a bit of a newbie post but we all had to start somewhere right ..
I'm wondering if someone can briefly explain the difference between blind and
attended transfers and why they would generate two very different CDR entries.
From my own research, it seems that transfers are both
> That's my question...the sbc provides security over trunking, right? The
> same can do Asterisk or a Proxy..isn't? Does an SBC can provide any kind of
> add-value to an Asterisk deployment?
A PBX provides functionality to users. An SBC *can* secure a PBX
against the outside world, but that is co
That's my question...the sbc provides security over trunking, right? The
same can do Asterisk or a Proxy..isn't? Does an SBC can provide any kind of
add-value to an Asterisk deployment?
El 11/06/2012 20:20, "Paul Belanger"
escribió:
> On 12-06-11 02:06 PM, Danny Dias wrote:
>
>> Hello,
>>
>> I wo
On 12-06-11 02:06 PM, Danny Dias wrote:
Hello,
I would like to know the difference between encrypt the rtp and signaling
between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm
trying to understand whether an SBC could fit an Asterisk deployment
What are you expecting the S
Hello,
I would like to know the difference between encrypt the rtp and signaling
between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm
trying to understand whether an SBC could fit an Asterisk deployment
Thanks
--
_
Hello,
Firstable, happy new year everybody! I have couple problems with asterisk
1.6.0.5 (meetme and ztdummy - kernel panic). And I want upgrade to newer
version of asterisk. I want ask you, what are the main differences between
these asterisk 1.6.1.x and 1.6.0.x branches? And which is better f
On Mon, Feb 11, 2008 at 05:25:44PM +0200, Khaled Chehab wrote:
> What are the differences between asterisk 1.2.4 and 1.4.6 beta
You probably ask about Asterisk 1.4 vs. Asterisk 1.6 beta, right?
>
> In functionality ,services
You can probably read about some of the changes in the file UPGRADE
Hi All
What are the differences between asterisk 1.2.4 and 1.4.6 beta
In functionality ,services and bugs.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail w
The release notes in the Subversion trees and bugs.digium.com will
probably serve to illuminate that.
Khaled Chehab wrote:
> What are the differences between asterisk 1.2.4 and 1.4.6 beta
>
> In functionality ,services and bugs.
>
>
>
>
>
> Regards
>
>
>
>
>
> -
What are the differences between asterisk 1.2.4 and 1.4.6 beta
In functionality ,services and bugs.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without
I'm rolling back from * version 1.2 to 1.0. Now I need to know what features
from 1.2 won't work on 1.0. So far I know this:
- features.conf doesn't work (does 1.0 have features.conf?)
- in extensions.conf I can't use
- as priority I can't use "n"
- Set(CALLERID(name)=)
-
Title: Differences between System 75 and Asterisk
I remember AT&T System 75 PBX systems back in the day and was amazed with how easily everything worked and was reconfigurable on the fly. Asterisk is also approximately the same. What are some of the differences between both units? Has anyo
--On Wednesday, September 22, 2004 14:06 -0400 Steve Kann
<[EMAIL PROTECTED]> wrote:
Try app_conference. In this configuration, you should be able to handle
200++ users without problems. It's ideal for this kind of thing.
(it's located in iaxclient CVS at iaxclient.sf.net).
Is there a link/WIK
Hello there
I've got kind of an odd problem. I have a setup with a lot of SIP channels
and a 30 channel PRI which is working perfectly.
In order to bill the customers I fetch cdr files from the PRI provider,
those files are generated every two hours. If I compare the CDR files from
Asterisk and t
28 matches
Mail list logo
