Hello folks,
since I started with asterisk 1.8.2 I got this messages in my console when
finish a call.
-- Executing [1610@from-e1:1] Dial(SIP/xxx-0027, SIP/1610,60) in new
stack
== Using SIP RTP CoS mark 5
-- Called 1610
-- SIP/1610-0028 is ringing
--
On 11-03-15 10:11 AM, Fellipe Paes wrote:
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x794f840 (len 828) to (null) returned -1: Invalid argument
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3386 retrans_pkt: Retransmission
timeout reached on transmission
On 03/15/2011 10:18 AM, Paul Belanger wrote:
On 11-03-15 10:11 AM, Fellipe Paes wrote:
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x794f840 (len 828) to (null) returned -1: Invalid argument
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3386 retrans_pkt:
: Re: [asterisk-users] Some errors
On 11-03-15 10:11 AM, Fellipe Paes wrote:
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x794f840 (len 828) to (null) returned -1: Invalid argument
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3386 retrans_pkt:
Retransmission
On 3/15/2011 11:18 AM, Paul Belanger wrote:
Theses are leftover issue with the IPv6 conversion for Asterisk 1.8.
Collect a complete debug log[1] and open a new issue on the tracker.
I believe one was entered a few months ago-
https://issues.asterisk.org/view.php?id=18514
--
Jeremy Kister
On 11-03-15 12:51 PM, Fellipe Paes wrote:
thanks for your answer, I'll open this issue on the tracker, but now I have a
new question, is there some way to deactivate IPv6 on Asterisk 1.8?
You can enable / disable IPv6 via the config files, but the core API /
ABI have changed.
But Kevin is
the reason you're seeing this, is your usage of _. as an
extension. Never do that.
-
Again, thanks for all.
Best regards,
Fellipe
Date: Tue, 15 Mar 2011 13:19:44 -0400
From: pabelan...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Some
On 03/15/2011 12:34 PM, Fellipe Paes wrote:
why I can't use _. in my dialplan?
Because it matches everything. In this case, it's matching the 'h' exten. So
when the call gets hung up, it goes to _. and does what you're seeing.
--
Hi Jason,
well, I was thinking something like this, but don't hurt to ask :D
Thank you for all guys.
Best regards,
Fellipe
Date: Tue, 15 Mar 2011 12:37:05 -0500
From: jpar...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Some errors
On 03/15/2011 12:34 PM
()
Thanks for all.
Best regards,
Fellipe
From: fellipe...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 15 Mar 2011 18:14:45 +
Subject: Re: [asterisk-users] Some errors
Hi Jason,
well, I was thinking something like this, but don't hurt to ask :D
Thank you for all guys.
Best
-users@lists.digium.com
Date: Tue, 15 Mar 2011 18:49:32 +
Subject: Re: [asterisk-users] Some errors
Hello guys,
one more question, if I have the following dialplan and can't use _. how can I
send everything that isn't 1620,1622,1610,9XXX to SIP/xxx?
Sorry I'm new with this * world.
root
On 11-03-15 04:18 PM, Fellipe Paes wrote:
I've solved this question just adding X in dialplan and killing that h option,
and of course with your help:
Since you are just learning Asterisk, I would *highly* recommend not
using 'exten = _X.'; this is a bad practice. Taking the time to
I have some problem to configure the call from asterisk to ser.[globals]SERADDRESS=xxx.xxx.xxx.xxx:5060exten = 77,1,Dial(SIP/[EMAIL PROTECTED],20,r)Error in Sip Debug ---NOTICE[25541]: chan_sip.c:6848 handle_response: Failed to authenticate on INVITE to
Hello Again,
As you said It may be the problem with CODEC which i configured in my SIP.CONF.
I used followoing code for CODEC in SIPCONF file :
disallow=all ; Disallow all codecsallow=gsm;allow=g723.1;allow=ulaw ; Allow codecs in order of preference;allow=alaw;allow=gsm;allow=ilbc
;allow=ilbc
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