Displaying 20 results from an estimated 1000 matches similar to: "codec translation problem???"
2006 Jan 27
0
How to put peers into Realtime
I have something like below in my sip.conf. How can I put this into
Real-time?
[voipbuster]
type=friend ; (or "peer" if we don't need incoming calls, or if there is
a separate section with "type=user")
host=sip1.voipbuster.com
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
username=abcd1 ;={{YOURUSERNAME}}
fromuser=abcd1
2006 Oct 24
0
sip.conf - srvlookup
I would like to put srvlookup=no in my SIP conf, so that I don't get DNS issues (Asterisk stops responding). I use VoIP Buster and in sip.conf I use sip1.voipbuster.com. When I do sip show peers in CLI I get
voipbuster/tomo 194.221.62.207 5060 OK (27 ms)
And when I ping sip1.voipbuster.com
[root@tomo ~]# ping sip1.voipbuster.com
PING sip1.voipbuster.com
2006 Jan 19
0
Incoming fax on voipbuster
Hello,
I'm trying to receive a fax to my inbound number from voipbuster.
Asterisk receives the call and starts the rxfax application successful,
but then nothing happens. The calling party is still hearing a ringing
tone, or sometimes nothing. Voicecalls are working correct and without
problems.
For testing I've add a local number (300) to the dialplan. When I call
this number
2006 Jun 08
0
"I can hear them but they can't hear me" with VoipBuster
Hi;?
When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly.
I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example.
?
I tried with different codecs: gsm, alaw and ulaw but no change.
?
So, now?I
2006 Jan 21
1
Is sip1.voipbuster.com corking reliably for others on list?
I am trying to move from IAX2 to SIP for voipbuster, moving at the same
time to sip1.voipbuster.com.
When I try calling out, I see that there is SIP exchange, and in many
cases also RTP data being exchanged.
Hover in a very large number of attempts the connection is not
established. Half of the time there is no RTP, the rest of the time there
*is* RTP data flowing in two ways, but no ringtone is
2006 Feb 20
2
spa3000
I'm trying to get working a spa3000 with asterisk. My problem is I
cant get wroking the incomming calls (I installed the lastest
firmware). My problem is asterisk is rejecting the authentication from
the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I
think I placed the username and password correctly...
Sip.conf says this:
[linea2]
username=linea2
type=peer
secret=1111
2014 Dec 23
0
Fwd: no ipv6 dns resolution for outbound registration with pjsip/asterisk13.1
3rd attempt to post it to the list, please ignore if it is duplicate
I have the following problem
When trying to setup asterisk 13.1 with PJSIP to connect to my IPV6 capable
SIP provider the registration fails.
[code][Dec 22 19:24:24] DEBUG[25247] pjsip: tsx0x110736c .Transaction
created for Request msg REGISTER/cseq=36181 (tdta0x721d90)
[Dec 22 19:24:24] DEBUG[25247] pjsip:
2004 Jul 30
1
SIP connections do not hang up
Hi everybody,
I have strange problem:
I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone
using Zap Channel) using sipgate to a number in public network.
When I'm hanging up before the other person picked up the phone, the line is
not closed correctly.
The phone keeps on ringing until timeout (of Sipgate I assume) and it even
costs my money, if the other person
2016 Oct 15
2
Registered successfully, but after a minute or so no SIP messages anymore
hi,
let me explain in detail, what i have configured and what is happening now:
1st router w724v (Deutsche Telekom AG):
- port forwarding, everything to destination port 51000-55999 to
device with ip 192.168.2.50 (interface of 2nd router)
2nd router Bintec RS353j):
- configured NAT, everything to port 51000-55999 to device
192.168.3.99 (same ports)
other direction is totally open.
I
2004 Oct 04
1
How to see CODEC which is in use?
How can I see which codec is in use during conversation. I can see (for
example) which codecs are negotiated before SIP connection, but I don't
know which is chosen:
12 headers, 12 lines
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 101
Peer audio RTP is at port 217.10.79.30:15666
Found description format GSM
Found description format iLBC
2012 Jun 18
1
Error SIP/2.0 488 Not acceptable here
Hello,
a person trying to call me by my phone number is getting the error 488 Not
acceptable here. I googled that error, seems like this error is normally
caused by a failed codec negotation, though I have no clue how I could have
read this out of the logs. Anyway, my setup is as follows:
Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider
The user calling me is also using Sipgate and is calling my
2005 Oct 10
1
Incoming SIP getting in, but not ringing.
Hi all.
Just as a quote note, can I thank everyone on this list. I find my
self finding pretty much every answer I am looking for on here. And a
big thanks to all thoughs helping us out. Mass Respect :)
Ok, I'm using a SIP provider (SipGate UK) to do my international
dialing etc, working great from extension 8 on phones. However some
more friends/contacts have started using SipGate also, and
2006 May 29
4
registration at Voipbuster times out
Hi,
I am new here on this list, and have a problem of which I hope that somebody here can help me with it.
I have a Voipbuster account, with which I would like to make phone calls via my Asterisk PBX. If I let X-Lite register directly at voipbuster.com, everything is OK, but if I let Asterisk register there, it says "registration for XXXXXX@sip.voipbuster.com timed out, trying again",
2005 Sep 10
2
VoipBuster again
Hi, all
I am still battling to connect * and voipbuster.
What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or
IAX traffic when using their client.
VoipBuster client connects to connectionserver.voipbuster.com on port 11112
for authentication. Call itself is placed on different server.
I have tried to connect using SIP and IAX and it seems that no
authentication is
2009 Sep 02
1
Voipbuster not ringing, other SIP peers are ringing...
Does anybody else see the same behavior for VoipBuster connections?
When I trace one of the other SIP peers, I see it sends this message:
----------------------------------------------------------------------
<--- SIP read from 82.101.62.99:5060 --->
SIP/2.0 180 Ringing
Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE
Call-ID: 740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl
2006 Jun 14
0
Strange problem with MusicOnHold - works outgoing - works with extension - but not incoming!
I've got a strange situation with musiconhold.
It works if I dial my extension 6000:
>From extensions.conf:
exten => 6000,1,Answer
exten => 6000,2,MusicOnHold()
Debug output if I call 6000:
-- Executing Answer("SIP/gs1-b6ee", "") in new stack
-- Executing MusicOnHold("SIP/gs1-b6ee", "") in new stack
-- Started music on hold,
2007 Mar 23
3
Semi-OT: Use T.38 ATAs to Extend fax lines
Greetings.
I have a scenario I would like some advice on. I have a 100,000 square
foot building that we will be moving some work crews into. It has
offices on each end of the building and a fiber line between them. I
currently have an asterisk 1.2 system in place and about 30 phones. My
problem is they want a few fax machines out in the warehouse area where
I currently have no wiring for
2006 Oct 17
0
lots of registrations, sip problem
Hello,
I've got a problem with connection to my SIP provider. In general,
everything works, but I get lots of these messages:
Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's
odd... Got a response on a call we dont know about. Cseq 42710 Cmd
SIP/2.0
Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request:
That's odd... Got a response on a call
2009 Jul 03
0
e164.org and tollfree ENUM records
Recently, I've been having issues with the URIs returned from e.164.org and
toll free calls. It seems that the URIs that are returned from ENUMQUERY and
ENUMRESULT are no longer the proper numbering schemes that the poviders use.
I've been using the following [enum] template in my outbound route for quite
some time with great success until recently.
[enum](!)
exten =>
2013 May 01
1
Call "stuck" in queue
Asterisk 11.1.0
One queue with strategy=leastrecent. (Full queues.conf below.)
Occasionally (several times today), a caller will get "stuck" in the
queue - there are operators available to take the call, but the caller
stays in the queue for a long time. Any idea what might cause this, or
where I can start looking to debug it? I'm going to start digging
through the queue log