[asterisk-users] Yealink T21P E2 bug solved

[Marcelo Terres]

https://www.mundoopensource.com.br/yealink-t21p_e2-com-bugs-no-firmware-bugs-in-the-yealink-t21p_e2-firmware/

[]s

Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

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Re: [asterisk-users] SIP trunk

[A J Stiles]

On Tuesday 26 Jul 2016, Jerry Geis wrote:
> It seems I am not getting any digits coming over a SIP trunk.
> 
> How can I match "anything" or "nothing" and start my extension.
> 
> Usually I have something like:
> exten => 55,1,Goto(,yyy,1)
> 
> but if 55 does not come across and it appears to be no digits
> coming across how do I match that that and just start.

Try using extension "s"  (for Start).  NB: Take care not to include another 
"s" extension into your context!

You can do something like
exten => s,1,NoOp(${EXTEN})
which will display in the console, whatever Asterisk thinks was dialled on the 
far end.  If a SIP trunk really is not sending any digits, it can only be used 
for a single incoming line; the calling party will have to send DTMF to select 
an extension once your Asterisk answers.


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AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] SIP trunk

[Tony Mountifield]

Hi Jerry,

In article ,
Jerry Geis  wrote:
> 
> It seems I am not getting any digits coming over a SIP trunk.
> 
> How can I match "anything" or "nothing" and start my extension.
> 
> Usually I have something like:
> exten => 55,1,Goto(,yyy,1)
> 
> but if 55 does not come across and it appears to be no digits
> coming across how do I match that that and just start.
> 
> I thought about _X but that says digits. I dont think I am getting any
> digits
> I just want *anything* coming across to start the call.
> 
> Basically ANY call coming across the trunk just do the same as 55 above.

It sounds like you are thinking of your SIP trunk as if it is a telephone
line, when you first pick up the line, and then send digits over it.

SIP doesn't work like that. A SIP call is started by an INVITE message
that says who the caller is, and what address they want to call. Normally,
the address would be something like sip:12...@some.realm.com, where the
12345 would be the digits of the number that would be matched against
your dialplan. It doesn't have to be, though, and you could have a call
to sip:je...@some.realm.com, which would match the following:

exten => jerry,1,Goto(and so on)

It is possible, but often not helpful, to have a call that doesn't specify
a number, with an address such as sip:some.realm.com, and that should
match the "s" extension:

exten => s,1,NoOp(Didn't get a number)

Maybe that's what is happening in your case, so try adding an "s" extension.

Hope this helps,

Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] SIP trunk

[Marcelo Terres]

_.  ?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On Tue, Jul 26, 2016 at 11:39 AM, Jerry Geis  wrote:
> It seems I am not getting any digits coming over a SIP trunk.
>
> How can I match "anything" or "nothing" and start my extension.
>
> Usually I have something like:
> exten => 55,1,Goto(,yyy,1)
>
> but if 55 does not come across and it appears to be no digits
> coming across how do I match that that and just start.
>
> I thought about _X but that says digits. I dont think I am getting any
> digits
> I just want *anything* coming across to start the call.
>
> Basically ANY call coming across the trunk just do the same as 55 above.
>
> Thanks,
>
> Jerry
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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[asterisk-users] SIP trunk

[Jerry Geis]

It seems I am not getting any digits coming over a SIP trunk.

How can I match "anything" or "nothing" and start my extension.

Usually I have something like:
exten => 55,1,Goto(,yyy,1)

but if 55 does not come across and it appears to be no digits
coming across how do I match that that and just start.

I thought about _X but that says digits. I dont think I am getting any
digits
I just want *anything* coming across to start the call.

Basically ANY call coming across the trunk just do the same as 55 above.

Thanks,

Jerry
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Re: [asterisk-users] PJSIP - Video Support for WebRTC

[Olivier]

Matthew Jordan  digium.com> writes:

> 
> On Mon, Mar 23, 2015 at 8:55 AM, Gosmac  gmail.com> 
wrote:
> > Hey i have an interesting topic to discuss here.
> >
> > The main goal here is to be able to make a video call between two 
WebRTC endpoints registered on asterisk 13
> it is a feature that definitely asterisk 13 should support .
> >
> > the problems that i faced with this is the following and i hope i 
could get an advise here.
> >
> > asterisk 13 vanilla version has some issues marking the video 
packets this complain web browser
> specially VP8 codecs so a friend of mine help me to patch 
res_rtp_asterisk and now asterisk is marking
> video streams :) it just mark video packets not touch anything else 
and web browser show video on web page
> now I’m using online demo http://tryit.jssip.net/ is stable and get 
more updates than sipml5. so i try
> echo() dialplan test and everything work perfect on echo test :).
> >
> > i have two questions and i hope you could give me some advise.
> >
> > 1) after marking video packet I’m able to make Dial() between two 
webrtc peers but i get one way audio and
> video on callee party, “after 3 minutes on call” i get two way audio 
and video on all parties seems to be
> not just a problem on a missing keyframe.
> >
> >  1.1) the 3 minutes delay only happen using chrome stable , could be 
a dtls problem when asterisk make an
> offer to other endpoint?
> >  1.2) when i use chrome-dev and i disable dlts encryption everything 
work perfect on video call.
> >
> > 2) after marking video packets i realize that when you make a call 
with video and you involve on dialplan an
> application like playback or music on hold any application that  
played audio files (audio and video never work).
> >
> > 2.1) asterisk is muggling the audio and video streams ?
> >
> > This is good information for all guys out there that wants to 
support video on webrtc in asterisk 13
> >
> 
> Please stop spamming the list with this e-mail. Resending it multiple
> times is clearly not yielding the results you'd like.
> 

Hi Matthew,
I'm testing WebRTC (JSSIP) with Asterisk 12.8 after following the 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support link. 
Using Firefox, I can connect both JSSIP Clients to asterisk. When I Call 
one Client, the Client just Ring One Time and after pick up a receive 
WebRTC error on the Firefox browser.
Here is my asterisk sip debug:

<--- SIP read from WS:192.168.2.103:49851 ---> INVITE 
sip:6000@192.168.2.106 SIP/2.0
Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK8394689
Max-Forwards: 69
To: 
From: "6001" ;tag=m6bqn333dr
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6407 INVITE
X-Can-Renegotiate: false
Contact: 
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: ice,replaces,outbound
User-Agent: JsSIP 2.0.2
Content-Length: 3158

v=0
o=mozilla...THIS_IS_SDPARTA-47.0.1 5760840281459352758 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256
E8:B7:2A:C2:DF:8B:AA:74:E6:D6:93:1C:68:88:81:39:82:C2:31:45:3D:8C:23:DF:
C1:23:72:03:F6:61:CC:F6
a=group:BUNDLE sdparta_0 sdparta_1
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 56808 UDP/TLS/RTP/SAVPF 109 9 0 8 c=IN IP4 87.169.189.102 
a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
56806 typ host
a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
56807 typ host
a=candidate:4 1 UDP 2122055935 192.168.2.103 56808 typ host
a=candidate:6 1 UDP 2122252543 192.168.56.1 56809 typ host a=candidate:0 
2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a
56810 typ host
a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc:43fa
56811 typ host
a=candidate:4 2 UDP 2122055934 192.168.2.103 56812 typ host
a=candidate:6 2 UDP 2122252542 192.168.56.1 56813 typ host
a=candidate:5 1 UDP 1685856255 87.169.189.102 56808 typ srflx raddr
192.168.2.103 rport 56808
a=candidate:5 2 UDP 1685856254 87.169.189.102 56812 typ srflx raddr
192.168.2.103 rport 56812
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:109 maxplaybackrate=48000;stereo=1 a=ice-
pwd:138f583004cb3079134e8e8f20dac36f
a=ice-ufrag:0941ac54
a=mid:sdparta_0
a=msid:{fb724d76-44fe-4e7d-a8d8-e4c00b4b57fe}
{bba6da45-42c8-4529-8f4b-046cffcdc40d}
a=rtcp:56812 IN IP4 87.169.189.102
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:540714091 cname:{ecde75c0-993f-44af-b136-8944915fe31c}
m=video 56816 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 87.169.189.102 
a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
56814 typ host
a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
56815 typ host
a=candidate:4 1 UDP 2122055935 192.168.2.103 56816 typ host
a=candidate:6 1 UDP 2122252543 192.168.56.1 60290 typ host a=candidate:0 
2 UDP 2122187006 

[asterisk-users] __sip_xmit Returned -1 Invalid Argument

[Brian Wilson]

Reviving an old thread, still seeing this.

Brian Wilson wrote:
>* I've been getting slammed with these messages on my console lately.
*>>* ed -1: Invalid argument
*>* [2016-05-31 10:09:40] WARNING[16249]: chan_sip.c:3775 __sip_xmit:
*>* sip_xmit of 0x7f05140803f0 (len 559) to 192.168.1.45:0

*>* > returned -1:
Invalid argument
*>* [2016-05-31 10:09:40] WARNING[16249]: chan_sip.c:3775 __sip_xmit:
*>* sip_xmit of 0x7f05140803f0 (len 559) to 192.168.1.45:0

*>* > returned -1:
Invalid argument
*
Joshua Colp responded:
> It's an invalid argument because the port appears to be 0,
> which won't work. How it got to be that I don't know... a full log may provide
further enlightenment.

Currently I have 9000+ messages similar to this:

[2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 725) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 725) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 727) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 728) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 729) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:06:53] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 729) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:07:04] WARNING[14771] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 711) to 192.168.89.172:54499 returned -2: Success
[2016-07-25 23:22:33] WARNING[15801] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 711) to 192.168.89.172:54502 returned -2: Success
[2016-07-25 23:22:33] WARNING[15802] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 711) to 192.168.89.172:54502 returned -2: Success

Port is not zero on these. Error code has changed from -1 to -2

What do you mean by a "full log".

Thanks -- Brian
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