Displaying 20 results from an estimated 3000 matches similar to: "Asterisk does not reINVITE after 302Redirect & 401Unauthorized"
2007 Mar 29
2
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-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
asterisk-users-request@lists.digium.com
Sent: Thursday, March 29, 2007 9:14 AM
To: asterisk-users@lists.digium.com
Subject: asterisk-users Digest, Vol 32, Issue 118
Send asterisk-users
2012 Aug 18
1
asterisk tries reinvite when incompatible codecs on call legs
Hi,
I just ran into what seems to be an issue on re-invites. I'm not sure if
it's a bug or as designed, so I thought I'd ask the question.
Here's my setup:
- Asterisk 1.8.13.0
- Phone A: Polycom ip331, only allowed to use ulaw, canreinvite=yes
- Phone B: Polycom ip330, only allowed to use alaw, canreinvite=yes
Phone A calls the extension of phone B.
After the normal call setup
2007 Jul 06
3
Mushtaq Ahmed is out of the office.
I will be out of the office starting 07/06/2007 and will not return until
07/09/2007.
For anything urgent pls. contact Fritz Reeve.
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2003 Jun 09
2
XML package for R
Hi,
Does anyone have a binary of XML package for 1.7.0 on Windows?
I have searched it at the RSXML pages (http://www.omegahat.org/RSXML/ ) but it seems it is no longer support.
Any pointers for the compiled version will be appreciated.
Regards,
mushtaq
[[alternate HTML version deleted]]
2007 Jul 07
2
Corporate Feedback to OSS (was: Re: Mushtaq Ahmed is out of the office.)
On Sat, 2007-07-07 at 08:39 -0500,
asterisk-users-request at lists.digium.com wrote:
> Date: Fri, 06 Jul 2007 12:02:53 -0600
> From: Stephen Bosch <posting at vodacomm.ca>
> Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the office.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
2005 Sep 15
2
SIP reinvite asterisk and NAT
I would like to setup up a remote office with a half dozen or so SIP
phones connected to an asterisk server via a WAN link. To conserve
bandwidth I would like the phones to be able to re-invite when they call
each other.
The phones will be Polycom, Cisco, or Snom.
I may or may not use NAT. Seems like the NAT would really mess up
re-invites, any experience with that?
Assuming no NAT,
2006 Jun 12
2
No reinvite - reason?
Hi,
I put reinvite=yes in my sip.conf.
For testing, I restricted the codecs to alaw.
I have no modifiers in my dial command.
Thus, there should be no reason not to reinvite.
Call (sip, authenticated) comes in and is forward
via SIP (not authenticated) to another asterisk box.
Unfortunately, media path still passes through the asterisk
box in the middle.
Using sip debug I even can't find
2004 Oct 07
2
Asterisk ---- SER ----- GAteway and Reinvite
Hi,
i'm using * with SER and a cisco 3725 as Gateway.
I noticed that the reinvite is not working if i use SER and if i don't use IT
(*---->Gateway) the reivite works so the * server is able to let the RTP
direct from gateway to SIP Clients.
Do you know in which way can i let it work with the SER too.
Becouse i need SER to manage other VOIP communities but if i'm not able to use
2009 Jul 21
0
Audio lost on reinvite
Hello, all. We are having a problem where audio for sip channels is
dropping upon reinvite. Perhaps it reflects a misunderstanding of what
reinvite does. We are running Asterisk 1.6.1.1 on CentOS 5.3.
SIP is set to canreinvite=nonat. We have tried RTP with strictrtp set
to both yes and no. We have also tried extending the Asterisk rtp port
range to accommodate the differing default ranges of
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
2009 Aug 02
1
T.38 and reinvite
I have a setup with a number of customer Asterisks with T.38 enabled.
This works quite well for each customer sending faxes between branch
offices.
They all have a SIP trunk to a central Asterisk, which connects them to
the PSTN through various providers on dedicated lines. I cannot enable
reinvite on those SIP trunks, because that would allow calls from the
customer's phones to get
2014 Mar 07
0
Problem with reINVITE on BYE
Hello all.
I am currently using Asterisk 11.7.0 (also tried Asterisk 12, but same
behavior) and is having an issue when it comes to reINVITE on BYEs.
Apparently one of the SIP providers that I am using does not always process
reINVITEs correctly, and would return a 500 Internal Server Error message
on some (but not all) of these transactions.
To get around this issue, I have been using
2006 Mar 07
1
Changing REINVITE status of the channel dynamically
I've an Asterisk server running in my office, which forwards all
long distance calls to a third party SIP service using an extension rule:
exten => _1XX0.,1,Dial(SIP/{EXTEN:4}@external_sip_server.com)
(1XX0 is the international calls rule for Chile)
Also, in my sip.conf, I've defined canreinvite=yes to decrease the
network load to the server caused by the RTP.
However, the external
2004 Aug 10
1
SIP Transfers (Possibly reinvite)
Hey Folks,
Is it possible to transfer an incoming call back out without a "trombone"
effect.
For instance;
Caller dials my broadvoice # --> Asterisk Answers and plays a menu --> the
caller selects an option --> asterisk transfers the call to my cell phone
via broadvoice and removes itself from the equation so I end up with...
Caller --> Broadvoice --> Cell Phone
Vs.
2009 Mar 16
1
T.38 - Which endpoint shall reINVITE ? caller or callee ?
Hi,
I've been playing with T.38.
I observed that mostly but not always, it's the "calling endpoint" that
reINVITE the other party to drop current SIP/G711 session and start a new
T.38.
But sometimes, it's also the callee party that reINVITE the calling party.
Which is the "standardized" or most common, way to start a T.38 session ?
Shall it come from callee or
2009 Feb 09
1
reinvite
I've never used "reinvite" in systems I have installed to date, and I have
finally run across a situation where it would be preferred.
A remote office has a flaky Internet connection. With G729 encoding the
calls to the central office over the 'net are tolerable. One Linksys 2102
drives two phones at this location, and when the first one calls the
second one it travels to
2006 Jun 06
1
Weird Can-Reinvite problem
Hi All,
I'm having a really weird can reinvite issue. I've been banging my head
around on this for days now..
I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5
172.20.0.11 is a hosted box and serves multiple offices
172.20.2.5 is a box on site at a customer's office.
A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phone
at 172.20.2.80 via server
2004 Jul 11
1
Stopping reinvite with IAX2?
Hi All,
I'm using DISA on my * server to avoid overseas toll charges when
making calls to Western Europe from my cell phone. I have DISA working
with a DID from a VoicePulse Connect account. The outgoing call to
Europe is also made via Voicepulse Connect.
I see that the IAX media path is bridging the inbound call to the
outbound call so that the media stream entirely bypasses my server once
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
2008 Jan 29
1
chanspy does not pull the call back to asterisk after a reinvite
Hello all,
I am allowing a reinvite between a snom 320 phone and a SIP gateway to take load off my Asterisk server. When I put the caller on hold, for example, Asterisk successfully reinserts itself into the rtp stream to play music on hold to the caller, but when I do a chanspy Asterisk does not seem to pull the call back. If I am spying on a channel when the call build up happens the reinvite