On 11 Mar 2015, at 17:53, Matthew Jordan mjor...@digium.com wrote:
On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes
steve-li...@geekinter.net wrote:
Anyone know where it’s gone?.. Appears to have been down all day.
The hamsters should be running in their wheels again now.
Cheers Matthew. Give
On Thu, 12 Mar 2015 12:52:46 +, Thufir wrote:
Heh, well, I guess it's dead:
http://www.digium.com/en/products/software/skype-for-asterisk
is this current?
http://www.remsys.com/blog/skype-connect-to-asterisk
it doesn't solve, I think, the problem I have that SIP
Your characterization may be true but Skype works much better than SIP
when it comes to sound quality.
I have SIP softphone with Asterisk server and Skype on the same
workstation.
Skype just works better over the same network.
Ron
On 12/03/2015 9:26 AM, A J Stiles wrote:
On Thursday 12 Mar
Hey all
We have been working with SIP for years. It has the potential to be better
than Skype. It is really all in the implementation.
Not all SIP soft clients are equal nor are the networks and computers they
are running on.
I will not bash Skype. We have tested it and in most cases
On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
Thank you, I needed a starting point to start my post.
1. Asterisk 12.8.1 (IAX2 voice issues) no video issues.
Voice issues on IAX2 Trunks, All extensions are SIP.
The IAX2 trunks on Asterisk 12.8.1
On Tue, Mar 10, 2015 at 6:11 PM, Chirag Desai djchill...@gmail.com wrote:
OK, it stopped working.
It turns out the transport and endpoints in PJSIP are ok. I can send an
invite from my unregistered snom phone and I can see some activity in the
CLI.
However, when I dial from my snom to
On Thursday 12 Mar 2015, Thufir wrote:
I'm testing Asterisk at home, crummy connection. Skype works fine for
me, but every SIP client, even without using Asterisk, fails to connect.
That's ok.
Is swapping out SIP for Skype a big deal?
Stay away from Skype! It is a toxic, proprietary
On Wed, Mar 11, 2015 at 7:11 PM, Jean-Denis Girard jd.gir...@sysnux.pf wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I made some tests with asterisk-13.2.0 and chan_pjsip this weekend
myself, and came to the same conclusion: some peerstatus events are
missing (eg. when contacts
On 3/12/15 9:39 AM, Ron Wheeler wrote:
Your characterization may be true but Skype works much better than SIP
when it comes to sound quality.
SIP is not to blame for this. Its the audio codec being used. Skype has
spend a great deal of effort with their SILK codec by making it highly
Which wideband codec did you use when testing SIP?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Thursday, March 12, 2015 9:39 AM
To: asterisk-users@lists.digium.com
Subject: Re:
On Thu, Mar 12, 2015 at 10:04:08AM -0400, Andres wrote:
On 3/12/15 9:39 AM, Ron Wheeler wrote:
Your characterization may be true but Skype works much better than
SIP when it comes to sound quality.
SIP is not to blame for this. Its the audio codec being used. Skype
has spend a great
On 3/12/15 12:19 PM, Administrator TOOTAI wrote:
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a écrit :
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},__dqs)
Here you have a problem: ${EXTEN} value is s
[...]
Daniel
D'Arcy J.M. Cain
If the device is registering and then dropping there are several usual
items.
The router may be closing the ports on the device.
The router may have a AGL SIP helper that is causing issues.
Make sure that the device is sending out keep alive packets.
Shut down any
2015-03-12 13:07 GMT-06:00 Bryant Zimmerman brya...@zktech.com:
SIPAddHeader(Alert-Info:\;info=ring3)
In the phone config add the value ring3 and select Account # / Call
Settings / Match Incoming Caller ID (Matching Rule)
In the first rule place the word ring3 and select your ring tone.
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = s,n,Hangup
app-chanspy]
exten = _0071XX,*1,*Macro(chanspy,1234)
exten = _0072XX,*1,*Macro(chanspy,5678)
exten = _0073XX,*1,*Macro(chanspy,8910)
but when i do
On Thu, 12 Mar 2015 08:58:01 +
Steven Howes steve-li...@geekinter.net wrote:
On 11 Mar 2015, at 17:53, Matthew Jordan mjor...@digium.com wrote:
On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes
steve-li...@geekinter.net wrote:
Anyone know where it’s gone?.. Appears to have been down all
On Thu, 12 Mar 2015 15:14:24 -0400
Bryant Zimmerman brya...@zktech.com wrote:
If the device is registering and then dropping there are several
usual items.
The router may be closing the ports on the device.
I don't see how. I am logged into the ATA through the router and I
don't lose the
This is driving me to distraction. I have a switch with multiple
clients who are all working fine except for one and I can't figure out
what makes them different. I have tried every NAT setting in the ATA
(SPA112 ATA with 2 x FXS, 1 x LAN), stun server on and off, different
sip ports, different
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a écrit :
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},__dqs)
Here you have a problem: ${EXTEN} value is s
[...]
Daniel
--
Hi list, someone has successfully change different ringtone from
dialpan with asterisk with this model Granstream?
for example:
exten = 0,1,Playback(pls-wait-connect-call)
same= n,SIPAddHeader(Alert-Info:;info=ring3)
same= n,Dial(SIP/310SIP/318,30,t)
can not get it to work
any idea o tips?
This is one of the drawbacks to using macros. There are workarounds for
macros, but the correct solution is use the Gosub / Return dialplan applications
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
SIPAddHeader(Alert-Info:\;info=ring3)
In the phone config add the value ring3 and select Account # / Call
Settings / Match Incoming Caller ID (Matching Rule)
In the first rule place the word ring3 and select your ring tone.
This will cause the selected ringtone to be used when calls
thank you but could you please tell me how can i put it
thanks and regards
2015-03-12 18:19 GMT+00:00 Administrator TOOTAI ad...@tootai.net:
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a écrit :
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten
I'm testing Asterisk at home, crummy connection. Skype works fine for
me, but every SIP client, even without using Asterisk, fails to connect.
That's ok.
Is swapping out SIP for Skype a big deal?
Heh, well, I guess it's dead:
http://www.digium.com/en/products/software/skype-for-asterisk
On Thu, Mar 12, 2015 at 5:11 PM, Chirag Desai djchill...@gmail.com wrote:
From: Matthew Jordan mjor...@digium.com
If the INVITE request is not shown in the CLI with 'pjsip set logger
on', then Asterisk is not actually receiving the request.
Does a pcap show the message being sent to
Followme is perfect to handle FMFM and it is now also realtime, but it
seems impossible to assign some value to a variable, from within the
followme to store info for example about the tenant the followme is running
under, like instead happen for example in the queue with the
setinterfacevar
From: Matthew Jordan mjor...@digium.com
If the INVITE request is not shown in the CLI with 'pjsip set logger
on', then Asterisk is not actually receiving the request.
Does a pcap show the message being sent to the correct IP/port? If you
change the transports to bind to port 5060, does
From: Matthew Jordan mjor...@digium.com
If the INVITE request is not shown in the CLI with 'pjsip set logger
on', then Asterisk is not actually receiving the request.
Does a pcap show the message being sent to the correct IP/port? If you
change the transports to bind to port 5060,
thank you so much it work
you must add 1 like below
[app-chanspy]
exten = _0071XX,*1,*Macro(chanspy,1234)
exten = _0072XX,*1,*Macro(chanspy,5678)
exten = _0073XX,*1,*Macro(chanspy,8910)
best regards.
2015-03-11 19:48 GMT+00:00 Carlos Chavez cur...@telecomabmex.com:
On 3/11/15 12:48 PM,
On Thu, Mar 12, 2015 at 5:14 PM, Leandro Dardini ldard...@gmail.com wrote:
Followme is perfect to handle FMFM and it is now also realtime, but it
seems impossible to assign some value to a variable, from within the
followme to store info for example about the tenant the followme is running
Hello,
Can anyone recommend a particular online WebRTC phone for testing with
Asterisk?
We tried:
- JsSIP, but even with the enable video checkbox disabled it sends video
options in the INVITE SDP and Asterisk rejects it with Rejecting secure
video stream without encryption details.
- sipML5,
Sipml5 works. You need to have TLS enabled on asterisk web socket.
Mitul
On 12-Mar-2015 12:47 PM, David Cunningham dcunning...@voisonics.com
wrote:
Hello,
Can anyone recommend a particular online WebRTC phone for testing with
Asterisk?
We tried:
- JsSIP, but even with the enable video
Hello David,
I'd recommend trying http://sipjs.com/ , it's similar to sipjs but you can
choose which kind of media it uses via a configuration object:
http://sipjs.com/guides/make-call/
Check out the guides, they are extremely clear and informative:
http://sipjs.com/guides/
cheers,
Olli
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