I’m setting up a new PBX in the Google cloud running FreePBX 14.0.1.36 /
Asterisk 14.7.5. Most calls are fine, but when calling an AT landline
that is busy, ringback tone is heard instead of the expected busy
signal. An example of a failing number is +1 408 269 1999 (a test number
that is
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
You can reload via http using a command like:
wget\
--output-document=/dev/null\
--quiet\
http://ip-address-of-pap/upgrade?http://ip-address-of-web-
server:80/asterisk/spa000F66A83C90.cfg
I tried it with my xml file and it complains about the file being
I am having an odd problem with a linksys pap2 ata and asterisk...
Asterisk won't detect digits from it until I issue a 'sip debug'. As
soon as I turn on sip debugging, everything works perfectly (classic
heisenbug)!
Instead of SIP debug, try capturing the traffic with tcpdump etc. on
the
Can I use my Linksys PAP2 with asterisk and an analog CLIP phone to
show the Caller number on the phone.
There's a Caller ID Method: option on Regional settings, but I
tested all options, and my CLIP phone never shows the Caller number...
It should work fine.
First, verify that you have for
I'm trying out a Linksys PAP2T-NA. Calling out works great, no problems
there. Calling in, though, the phone doesn't ring. Caller ID shows up, I
can pick up the phone, and the call is connected, but no ring. I've tried
it on two analog phones, same behavior. Suggestions?
I don't know if
Hi David and all,
I have a voip provider that uses mgcp and I would like to connect that
provider to my asterisk.
Anyone succeed in doing this?
I have a similar interest, for Free Télécom (France) DSL, which
includes an MGCP based VoIP service. I have been too lazy to tackle
this myself, but
Hi,
Are you sure that this is an Asterisk problem? Configure an IP phone,
ATA, or softphone to connect directly with the provider, and check the
quality. If it's bad, use tools such as
http://www.testyourvoip.com/ and
http://www.pingplotter.com/ to troubleshoot.
If standalone phone works ok,
Please note that recent IOS has SIP NAT traversal turned on by default.
I believe that it only supports internal UA / external server.
Since you also want the opposite, you should probably turn it off:
no ip nat service sip tcp port 5060
no ip nat service sip udp port 5060
Some IOS versions will
Hi Mike,
This is wanted because using to ATA back to back creates a number of
problems with echo. Also a delay for CID and problems with DTMF decoding.
Keep everything digital is the way to go.
Agreed. But before getting started with Asterisk, I posted a similar idea
to the group; it was met
mgcp.conf
[general]
port = 2427
bindaddr = 10.22.58.222
[10.22.58.199]
context=iad101e
host=dynamic
callerid = 169 169
nat=no
canreinvite=yes
line = aaln/0
extensions_additional.conf
exten = 169,1,Dial(MGCP/aaln/0 at 192.168.0.22)
now the problem is when i dial from a sip
Hi,
I have some SoundPoint IP 501 phones, running SIP 1.6.2.
I would like to configure them so that line 1 connects
directly to a SIP provider, and line 2 connects to a local
Asterisk PBX. That should be simple enough, but this
provider requires URIs like sip:[EMAIL PROTECTED] .
However, the DNS
Is there anyone out there who has given this outfit money and actually
received any service from them?
I am about to give up on sixTel, because of poor customer service.
It's a shame, because they otherwise seem quite competent.
I signed up Sept. 23 at http://www.iax.cc . Outbound service
I have callwaiting=yes in my zapata.conf, and Call Waiting Serv: Yes
in the PAP2s.
However, there's sitll no callwaiting on the PAP2s. Everything else work
fine. Any ideas? Am I missing something somewhere?
Hi Andy,
You also need to set CW Setting: Yes on the User 1 and User 2 screens.
Or,
im trying to make two asterisk boxes communicate on mgcp protocol only.
Anybody has idea how to implement this
This is presently not possible, unless you have some
suitable intermediate gateway(s).
MGCP is a master-slave protocol. The Call Agents
control the Media Gateways. The current
Whenever we call IBM, the call counter on the phone never starts and in
the CLI the zap channel never gets the answered signal from the PRI.
First, there is nothing unfair or illegal going on. Large toll-free
users have enough clout that they can negotiate contracts, where they
are not billed
Is there a way I can tell if it is asterisk or the carrier that is
timing out from the CLI?
Sorry, I don't have PRI and don't know the details.
However, I'm sure that if you set a high enough verbose or debug
level, you'll see the ISDN messages between * and the carrier's
switch. I don't know
Hi Paul,
I'm receiving the following error over and over, adnauseam:
Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce received from 'CNAME-CID
sip:[EMAIL PROTECTED]'
Does anyone know what stale nonce is?
Thanks!
This is normally not an error.
Digest authentication in
I'd like to use the VoIP service from Free with Asterisk,
but am having a couple of problems. Here are some details:
ADSL from Free Télécom comes bundled with VoIP and TV
services. Most users access the VoIP via the supplied
Freebox, which is an integrated DSL modem, router, ATA, and
media
Hi Luis,
Can anyone can tell me if I can connect 2 Cisco ATA-186 in a peer to peer
layout
(without an Asterisk server registerisng the devices) through Internet?
If running MGCP or SCCP, no.
If running H.323 or SIP, and both ATAs are on static public IPs, no problem.
Just specify the address
I am a little pissed when
all other ATA's are configurable from their built in web server.
The 2102 does have a built in Web server.
See manuals at support.bctgroup.ru/mediatrix/2102/
If you have a refurbished unit, perhaps the web server was disabled,
or the password was changed. Try reset to
The 2102 does have a built in Web server.
If you have a refurbished unit, perhaps the web server was disabled,
or the password was changed. Try reset to factory settings.
It is possible to disable the factory reset, and conceivable that
the previous owner did that. However, if he did, the SNMP
I am interested in implementing RTP over TCP
Why? If you want to permit operation through a firewall
that blocks UDP, there are packages that provide VPN
tunnels over TCP or even HTTP. You could then run
any VoIP system over that VPN. As you said, delay
performance would sometimes be awful.
Hi Ken,
Can't seem to find it anywhere, and my cisco login works, but says
there's no longer any downloads available for the ATA186.. anyone know
where I could find the MGCP version of the firmware via download?
Log in. From the main page, click the dropdown list for
Downloads and select
I've been there.. the page comes up with There are currently no files
for this type.
Well, you either have a technical problem or an administrative one.
Eliminate the possibility of corrupted cookies or browser cache by
going to another workstation, accessing
I've tried using iaxtel and BroadVoice to route toll free calls and the
call appears to connect ok (see log snippet below) but it just rings and
rings and eventually it times out and I get
The person you are calling is unavailable
Hi Shadow,
This is a common problem, not limited to
The next step would to be turn pedantic=yes back on, then generate a
failing call with 'sip debug', 'set verbose 255' and 'set debug 255' in
place. Capture all the output (there will be a lot) and then post a bug
in Mantis describing the situation and attaching the output file.
Kevin, thanks
Is it worth posting such a vague bug report? Unfortunately, I know
absolutely nothing about the internals of Asterisk.
Yes, please do, but make sure you include a full 'sip debug/set verbose
255/set debug 255' as an attachment in the bug. Also include the
relevant portions of your
I'm running [EMAIL PROTECTED] (Asterisk 1.0). Is it possible that this bug
has already been fixed in a later version (I can't find anything that
seems relevant at bugs.digium.com)?
This issue (multiple c= lines) has already been fixed in CVS HEAD (if
'pedantic' SIP parsing is enabled), but
Hi,
I'm testing Asterisk with a new provider. On calls to US
toll-free numbers, there is no audio (calls to normal numbers
are ok).
In response to a valid INVITE from Asterisk, something like
this is received:
SIP/2.0 183 Session Progress
v:SIP/2.0/UDP [my public
Hi,
I'm looking for a reliable, reasonably-priced, single-channel
interface between * and US GSM.
The VOIP GSM Gateways listed at
http://www.voip-info.org/wiki-VOIP+GSM+Gateways
(VoiceBlue, QUTEX) are multichannel systems, very expensive
($2500 or more).
Next step down, there are various Fixed
As far as I can make out the root password for the ISO download is
supposed to be epping or EPPING depending upon which version you are
using.
I've downloaded an ISO image from the following link but neither passwords
seem to work :(
Sam In France, the second most important ADSL provider (named Free)
Sam offers a phone line (which uses VoIP but can only be used as a FXS)
Sam with unlimited free calls to landlines.
I also have Free ADSL in Paris, and would very much like to get
their VoIP working natively with Asterisk. Free
The MOS (Mean Opinion Score) scale is:
5=Excellent; 4=Good; 3=Fair; 2=Poor; 1=Bad.
Some values, taken from Carrier Grade Voice over IP by
Daniel Collins:
G.711 4.3
G.729 4.0
G.729AB3.9
GSM(full rate) 3.7
The above scores assume no packet loss, minimal delay, no echo.
Has anyone tried SunRocket with Asterisk?
http://www.sunrocket.com/
The $199/yr. plan seems like an excellent value,
and most reviews have been favorable.
However, I don't know if it is possible to obtain the SIP
credentials, so one can bypass their gizmo.
Thanks,
Stewart
When I make a call to either 706 or 707 from any phone, the phone
attached to the spa does not ring. However, if I pick up the appropriate
phone, the connection is made and normal conversation can take place.
I had the same problem with a Cisco 827-4V. It turned out that the
phones were fussy
Is NAT enabled by default on Fedora core 1 (latest patches) ?
Sorry, don't know. I believe that if you have disabled iptables
by e.g. /etc/init.d/iptables stop
then NAT should be off, but it still wouldn't hurt to check the
source address reaching the phones.
The target machines can be pinged
I have a * box with 2 nics in the following setup:
Internet
|
192.168.5.253 (firewall)
|
192.168.5.xxx network (gw 192.168.5.253)
|
192.168.5.10 (* nic 1)
192.168.6.10 (* nic 2)
|
192.168.6.xxx network
The netmask for both networks is 255.255.255.0
The 192.168.6.xxx networks has
However, even though I've added the 192.168.6.10 as the gw
for the 192.168.6.xx network, the phones cannot access
the 192.168.5.xx network (or the internet).
Well, if you can open a TCP connection from 192.168.5.xx to
192.168.6.xx, then routing in the reverse direction must be
working. If you
Hi Rodney,
I dont have a cisco acount yet
can some bady hel me with the
ata18x-v2-16-030401a-1.zip file.
You will need a PC running Windows.
1. Unzip it.
2. Read the text file ata186us.txt
3. Follow instructions in it :)
This will convert your ATA from MGCP/SCCP to H.323/SIP .
--Stewart
Hi Rodney,
I dont have a cisco acount yet
can some bady hel me with the
ata18x-v2-16-030401a-1.zip file.
ftp://ftp.rekom.ru/pub/ata18x/
You will need a PC running Windows.
1. Unzip it.
2. Read the text file ata186us.txt
3. Follow instructions in it :)
This will convert your ATA from
I haven't found any recent information on this, but can Asterisk
act as a MGCP UserAgent?
I wish I was wrong, but I think right now MGCP in Asterisk is CallAgent
only.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20MGCP%20channels
Any other ideas for interacting with an MGCP
are they really /unlimited/ in the truest sense of the word ?
US$24.95, even if it's only for unlimited calls to Malaysia
(where i am) seems very, very attractive. when something is this
attractive, i start looking for the catch.
AFAIK, no one offers truly unlimited service. Companies
Anyways, found an unsecured wireless network going through my new townhouse
at 30% strength. Found the owner and they said I could share it for a couple
of weeks.
They have a Netgear, 108mbs 802.11 b/g. So I took a LinkSys WAP54g and put
it in Ethernet bridge mode, it took the signal and
Can you expand ... I'm hoping to use Asterisk to do SIP to MGCP
mediation ... what does it not work?
I don't know the particulars, because I've never used (or even looked at
MGCP). All I know is that whenever the issue comes up, people here say
that Asterisk does not know how to act as an MGCP
i have a audio problem between sip and h323.
First my installation:
Debian Sarge
Asterisk 1.0.1
Gnugk 2.0.8
Asterisk register a prefix to gnugk.
Communication from sip to sip and h323 to h323 is working.
When i now call from the siphone (three tested) the h323 phone (also
three tested) the
i have a audio problem between sip and h323.
First my installation:
Debian Sarge
Asterisk 1.0.1
Gnugk 2.0.8
Asterisk register a prefix to gnugk.
Communication from sip to sip and h323 to h323 is working.
When i now call from the siphone (three tested) the h323 phone (also
three tested) the
On the client side, I'm not sure
what the risk is to say a SIP phone that has 5060 and some rtp ports
forwarded to it. Maybe someone can come in and list the threats to
both ends of a double NAT setup? I'm sure hundreds of us would be very
interested in this!
Here is a simple example. A user with
I played around for a few hours with a polycom 500 phone and it seems me that the dtmf
mode is not configurable, looks like it only has inband mode.
While this is ok with G711 I assume that will result in some troubles
using G729, altought I cant test it because I havent got any g729 licence
yet.
I have a single analog line coming into the house.. This line
is for my
ADSL and home phone.. My Asterisk box uses an X100P card to
connect to
the analog line.. I have a microfilter on the line etc.. The
rest of my
phone system works inbound and outbound calls via a VoIP
provider over
the
Jeff,
I did a cut-n-paste of your configuration straight into my sip.conf,
updated the username and password. Still getting the same result as
before, audio in only one direction. Can can call between my local
SIP extensions fine, so I know my sipura box is working and configured
correctly.
I have tried another microfilter, the long cable and the cascaded
microfilter and all made no difference at all..
I dont think it is the microfilter or the internal house cabling.. Also
the fact that a standard analog phone doesn't do it also points to the
X100P..
I can't move the X100P to
Hi Benjamin,
I looked at NuFone.net and some others, but it appears that
IAX is not right for my system.
I'd say this is only because you don't know enough about IAX yet ;-)
[Many comments explaining how IAX would work wonderfully if all my
VoIP hardware were replaced with IAX-compatible
I presently have a small VoIP network using H.323 and gnugk,
and would like to upgrade it to an Asterisk-based system,
primarily to take advantage of low cost unlimited calling
plans offered by SIP providers such as Vonage. However, the
carriers with good reputations for reliability and quality
At first I thought the X100P was what I was looking for, but now it
looks to me like the X100P does not have an IP interface, so it would
require all audio to run through the CPU. I'm familiar with ATA186's,
which I think are comparable to the IAXy box, and I'd just like to find
something like
please take a look at these units:
http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1048701CatId=1596
The price of $30 after rebate certainly looks interesting.
are they locked?
If the firmware agrees with the manual at
http://www.voip2.net/Operator_Manual.pdf ,
it's not
Thanks for the replies. I do have vonage phone service and they have
provided me a motorla device I plug into my broadband and also plug my
phone into to make calls. this is a nice service for 30 bucks, but as
with all things linux, why cant one connect to the PSTN for free?
I suspect that
Could someone please recommend a reasonably priced IP phone
that works well with *, has a decent (full duplex, echo canceling)
speakerphone, has at least two line appearances, and can
transfer / conference reliably?
The Wiki lists 35 brands of hardphone, but:
1. Most seem to be toys.
2. For many,
Does anyone know a good (and stable) voip gateway product with 4 ports
(2 fxo and 2 fxs), with the following requirements:
* being able to connect analog phones to the FXS ports, and communicate
over SIP with an REGISTRAR/PROXY server (SER in our case).
* being able to connect the FXO port
work. It
should not be difficult to use Ethereal to see where the audio
is getting lost.
Regards,
Stewart
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: Stewart Nelson [EMAIL PROTECTED]
Sent: Monday, June 21, 2004 11:13 PM
Subject: Re: [Asterisk-Users] 183
Hi Philip,
Unfortunately, * speaks MGCP only as the Call Agent, rather
than as the Media Gateway. MGCP is a master/slave protocol,
and it would take some effort to make * work as the slave.
I have the same problem: Free Telecom here in Paris includes
MGCP service with their DSL. You can call
Hi Charles,
Blocking the 183 is undesirable, because messages
from the PSTN indicating that e.g. a number has been changed,
will be lost. Instead, do what's necessary to get audio
back to the caller. On the ATA, set bit 19 of ConnectMode
(see table 5-8 of manual). On the 5300, see
Hi,
You need to set the DialPlan parameter to allow the proper
number of digits to be collected, for all types of numbers
used in your system. I believe that the factory default
value would work for long numbers beginning 0011, but your unit
was probably previously configured for a different
Hi Martin,
This looks like a SIP reply.
I suspect that a misconfigured SIP phone or proxy is inserting
a Via: header that contains the 195.77 address, or a name that
resolves to it. Capture the packet text with your firewall,
or by running Ethereal on your * machine, or with * itself,
and the
Hi Matt,
On the ATA, set TxCodec=2 and RxCodec=2 (G.711u).
Also, set AudioMode=0x00160016 , which will force G.711 .
After saving, reload the /dev page to be sure that these
values are set as expected.
In Asterisk, allow=ulaw only.
If it still doesn't work, use the NPrintf field and
prserv,
Hi all,
I am looking for a software package (free or not), or an
inexpensive hardware device, which can route calls between
an H.323 network and an MGCP-based voice service.
Unfortunately, I believe (based on documentation and other forum
posts -- I have not looked at the code) that Asterisk can
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