Hey,
I've found a bit of chatter about people using hacks to copy voicemail
messages into object storage (like S3) after they've been recorded. But
I was wondering if any work has been done on the VoiceMail app to
actually store and retrieve messages to/from an object store?
Cheers,
Andrew
--
Nope, there are no contacts to show that pertain to these endpoints (only
my SIP trunks show up).
On Mon, Feb 15, 2016 at 5:31 PM, Joshua Colp wrote:
> Sonny Rajagopalan wrote:
>
>> Does this help:
>>
>
> Yes, the transport parameter is in the Contact header so it's
Does this help:
Session Initiation Protocol (REGISTER)
Request-Line: REGISTER sip:1.2.3.4;transport=TCP SIP/2.0
Method: REGISTER
Request-URI: sip:1.2.3.4;transport=TCP
Request-URI Host Part: 1.2.3.4
[Resent Packet: False]
Message Header
Via:
Sonny Rajagopalan wrote:
Does this help:
Yes, the transport parameter is in the Contact header so it's
interesting it didn't work. If you use pjsip show contacts what is the
contact for the AOR?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL
Carlos Chavez wrote:
Ok, thank you. So, if for example I have my main transport configured as:
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.2.0/24
external_media_address=XXX.XXX.XXX.XXX
external_signaling_address=XXX.XXX.XXX.XXX
Does that mean that I do not
On 2/15/16 1:08 PM, Joshua Colp wrote:
Carlos Chavez wrote:
On 2/15/16 12:50 PM, Joshua Colp wrote:
Carlos Chavez wrote:
Is it possible to use serveral protocols for a single transport
section
in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you
cound use webrtc along with
Sonny Rajagopalan wrote:
Thanks for the mighty quick response, Joshua!
I am using Zoiper on Linux softclient:
REGISTER sip:;transport=TCP SIP/2.0
That's the request URI, not the Contact header. The Contact contains the
URI that the server should dial to reach the client. The full message
Thanks for the mighty quick response, Joshua!
I am using Zoiper on Linux softclient:
REGISTER sip:;transport=TCP SIP/2.0
Changed the port back to 5060.
On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp wrote:
> Sonny Rajagopalan wrote:
>
>
>
>
> *CLI> pjsip set logger on
>>
Carlos Chavez wrote:
On 2/15/16 12:50 PM, Joshua Colp wrote:
Carlos Chavez wrote:
Is it possible to use serveral protocols for a single transport section
in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you
cound use webrtc along with your phones but if I try:
[transport-udp]
On 2/15/16 12:50 PM, Joshua Colp wrote:
Carlos Chavez wrote:
Is it possible to use serveral protocols for a single transport section
in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you
cound use webrtc along with your phones but if I try:
[transport-udp]
type=transport
Carlos Chavez wrote:
Is it possible to use serveral protocols for a single transport section
in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you
cound use webrtc along with your phones but if I try:
[transport-udp]
type=transport
protocol=udp,ws,wss
bind=0.0.0.0
No, each
Is it possible to use serveral protocols for a single transport
section in pjsip.con? In sip.conf you could use transport=udp,ws,wss so
you cound use webrtc along with your phones but if I try:
[transport-udp]
type=transport
protocol=udp,ws,wss
bind=0.0.0.0
I get an error that
Sonny Rajagopalan wrote:
*CLI> pjsip set logger on
PJSIP Logging enabled
[Feb 15 18:28:12] WARNING[30075]: pjsip:0 : tsx0x7f14b0003 .Temporary
failure in sending Request msg OPTIONS/cseq=52002 (tdta0x7f14b0001470),
will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
[Feb 15
This question was asked by Chirag on March 4 2015 earlier, but I am
following exactly the same procedure here and I cannot even get my clients
to register on Asterisk.
Here's my PJSIP.conf:
[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:5061
...
[endpoint_internal](!)
type=endpoint
On Mon, Feb 15, 2016 at 02:15:58PM +, Ryan, Travis wrote:
> Getting the some errors making dahdi 2.11.0.
>
> Seems same as listed here http://forums.asterisk.org/viewtopic.php?f=1=96455
>
> In that link they say to use 2.10.2 but that's from December. Is there a fix
> yet for this?
My bad,
Getting the some errors making dahdi 2.11.0.
Seems same as listed here http://forums.asterisk.org/viewtopic.php?f=1=96455
In that link they say to use 2.10.2 but that's from December. Is there a fix
yet for this?
Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North
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