Displaying 20 results from an estimated 10000 matches similar to: "Re Invite not working"
2005 Sep 05
0
ReInvite not working
Hi
Although canreinvite option is yes, the asterix doesn't send reinvite and the media is going through the asterix instead of between the two sip phones.
Both sip phones (handytone 486) don't use NAT and are configure with canreinvite option yes and use the same codec G.729. And Dial() command don't contains t or T.
Any suggestion on what could be the problem ?
2006 Mar 30
0
Strange second REINVITE being sent
I'm using Asterisk a SIP Server for a lot of GrandStream HandyTone
ATA's. Each one of them is configured in sip.conf as:
[1234567]
type=friend
username=1234567
secret=1234567
callerid="ATA 1234567"
host=dynamic
nat=yes
qualify=yes
disallow=all
allow=g729
canreinvite is set globally to YES.
When one ATA calls another, I see the next traffic on Ethereal (just
shown the sequence
2003 Nov 25
3
Handytone 286 - calling out
Hi,
Just received recently released Grandstream handytone 286 ATA for
testing.
I can call ATA from any other extensions and conversations seems to be
of quite good quality. However placing calls from ATA is not possible at
all to any extensions.
After dialing there no indications of any kind from ATA at all. It just
"hangs in there".
ATA is behind NAT, registers to an * with public IP
2016 Aug 10
2
Asterisk & Vitelity Invite issues
On 8/9/16 12:40 PM, Matt Fredrickson wrote:
> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote:
>> Hi All,
>>
>> We have asterisk 11.23 running sip to vitelity and from there IAX trunks
>> split off to where they need to go. We are having a problem getting
>> chan_sip to quit ignoring re-invites from Vitelity. Our side ends
2016 Aug 08
2
Asterisk & Vitelity Invite issues
Hi All,
We have asterisk 11.23 running sip to vitelity and from there IAX trunks
split off to where they need to go. We are having a problem getting
chan_sip to quit ignoring re-invites from Vitelity. Our side ends up
sending a reinvite which their side & they do not support us sending a
reinvite. Ive tried:
canreinvite=no which was supposedly replaced by:
directmedia=no
Can anyone shed
2004 Jan 04
8
Grandstream Handytone 286 RTP Problems
I am trying to get the handytone 286 to make a very simple call to * and
having problems. It registers with * just fine, but when I place a call
(to echo test, for example), the RTP stream seems to have problems
opening. Here is there error I get in *:
WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for
2006 Apr 10
0
Problem with Asterisk and Grandstream HT286
I've dealing with this issue for a while, and I'd really like to know if
anybody has experienced the same pain before :-)
I've a lot of Grandstream HandyTone 286, loaded with the latest firmware
(1.0.8.16) from the GS website. In my sip.conf, this ATA's are
configured as:
[05]
type=friend
username=05
secret=XXXX
callerid="User 05"
host=dynamic
nat=yes
qualify=yes
2014 May 12
1
new install: no re-invite and unwanted transcoding
I am unable to get re-invite to work on a new system. Also, unwanted
transcoding is occurring on PSTN calls.
The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will
eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2,
CentOS 5.8) currently in production. Both systems are on VPS with public
IP addresses. Goals for the new system include: HD (g722) connections on
2014 Dec 15
1
T.38 not working - help needed with log interpretation
On Mon, Dec 15, 2014 at 3:34 AM, Recursive <lists at binarus.de> wrote:
>
<snip>
>> For asterisk 1.6 & 1.8 you would need to set 'canreinvite=no', I don't know what Asterisk 13 will do with this setting.
>>
> I suspect Asterisk 13 will just ignore it. To make things worse, there seems to be a configuration directive named reinvite (not a typo); I
2004 Aug 19
0
SIP reinvite code negotiation
Hi,
We're routing SIP calls through Asterisk and we want to
be able to reinvite calls without Asterisk performing
codec conversion.
We've performed the following test:
Asterisk has license for G.729 installed
sip.conf
[general]
context=default
autocreatepeer=yes
disallow=all
allow=alaw
allow=g729
canreinvite=yes
nat=no
We have configured two endpoints:
EP1, preferred codec order
2015 May 15
1
Re-INVITE and bridge breakage
Hello,
as a variation of our issues with Adhearsion calls dropping when an INVITE
comes in for a bridged call, I now have a new issue to contend with.
Our call is in an AsyncAGI application, and has been bridged to another
channel.
The provider that supplies the DID sends a polling reINVITE every 15
minutes (it's a documented Metaswitch behavior amongst others).
The reINVITE is seen as a new
2004 Jul 19
0
(Asterisk-Users] Affordable SIP Phone - Stiil a Myth?
Steve,
Here is the config, I pulled from my server, that works with D'Link Phones:
Main Menu
--------------------------------------------------------------------------------
SIP.CONF
[general]
port = 5060 ; Port to bind to (SIP is 5060)
;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
bindaddr = 67.109.153.236
disallow=all
;allow=ilbc
allow=gsm
allow=ulaw
2005 Jun 03
0
SIP_CODEC, reinvites, and changing codecs
I am wondering if the SIP protocol and its implementation in * allows for
changing codecs mid-connection.
I've seen some questions regarding this on the list, but I've not found any
clear answers.
I've also seen the SIP_CODEC variable, but it's not clear that it will change
the codec on an existing call. Also, there are mentions of needing a reinvite
to make the change, but most
2005 Feb 16
1
Passthrough and reInvite
It is not clear how exactly g729 pass-through can be enabled. I
have a SIP call off a gateway come into an Asterisk menu, and then I
send the SIP call to another SIP gateway using Dial(). Even though
codec preferences have g729 listed first, it never gets used.
Both gateways have separate peer entries in sip.conf, and both have
canreinvite=yes set. Can Asterisk change the media type during
2007 May 19
1
asterisk not sending ACK after reinvite
Hi,
I am faced with this dilema of asterisk not sending an ACK after it receives
200 OK from OpenSER (which is a response to a reinvite request sent by
asterisk. Here is my setup
Carrier<->OpenSER<->Asterisk1<->Asterisk2
A user is connected with Asterisk1 (through the carrier and OpenSER). On
certain dtmf events the call is forwarded to Asterisk2 using the Dial
command.
2003 Oct 02
3
SIP and DSL Bandwidth queries.
Here is my setup
7960(A)--Firewall/PAT--dsl---------Internet--------dsl--Firewall/NAT---7960(B)
| |
| |
7960(C)--NAT--cable----------------- -----dsl -- Asterisk
(A) can communicate with (C) only when C is configured with canreinvite=no. The
call gets dropped in few seconds if canreinvite is set to yes for C.
2006 Dec 15
1
Cisco Call Manager 4.0 to Asterisk, Anyone have SIP Reinvite working?
Hi All,
I haven't started sip traces or debug yet, but was wondering what the deal
is with the CCM and reinvite, why it doesn't work with Asterisk (using
1.2.9.1). I can make calls back and forth all day with canreinvite=no, when
I try to reinvite an inbound sip call from the CCM with Asterisk server 1 to
Asterisk Server 2, I get one-way audio issues. All the RTP ports are
configured
2016 Nov 30
2
Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached
Hello all!
I can see a strange problem during invite in dialog in the context of
timer handling.
Given is the following incoming call from provider at 8.195.88.234 (2 at 2)
to my asterisk at 28.19.57.152 (1 at 1):
After 900s suddenly *asterisk* starts the timer reinvite - I would have
expected the reinvite started by the provider as usual.
The expected reinvite by the provider is started
2003 May 17
0
Debug for SIP and reINVITES (ATA-186)
I must be doing something incorrectly, or something is wrong with
ATA-186 reINVITEs in SIP. Perhaps someone more enlightened than me
can lend me a hand.
I have been attempting to get two SIP phones to reINVITE to each
other, and I've been unable to think of or uncover the correct
method. The calls always go through the Asterisk server, no matter
what I try. I've simplified things
2019 Aug 15
4
PJSIP reInvite
Hi All,
We are using asterisk 16.5 and having an issue with the first re-invite
after the call has been established.
We can see the call gets up and you see in the logs the bridge type has
changed and after that a re-invite is triggered.
Is there any possibility to deactivate this kind of reInvite? We have some
race conditions while have multiple asterisk in the call flow and the
different