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&T 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 alw
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
> You can reload via http using a command like:
> wget\
> --output-document=/dev/null\
> --quiet\
> "http:///upgrade?http:// server>:80/asterisk/spa000F66A83C90.cfg"
> I tried it with my xml file and it complains about the file being corrupt.
> I'm guessing you n
> 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
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 the
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, b
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, c
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 ev
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 m
> 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
> 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 w
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
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.
O
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 w
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 bill
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 versio
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
'
Does anyone know what "stale nonce" is?
Thanks!
This is normally not an error.
Digest authentication in SIP is very similar t
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 play
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
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 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 t
> 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.
> 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
http://www.cisco.com/cgi-bin/t
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 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
>> 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 s
> 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, than
>> 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 enable
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 IP]:5060;branch=z9hG4bK62d91cea
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 C
> 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 :(
>
http://ovh.dl.sourceforge.net:80/sourceforge/asteri
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. F
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.
How
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 a
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
> 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 pinge
> 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 y
> 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.x
>>> 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 a
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 M
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 .
--Stewar
> 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 di
In what context will Asterisk will require proxying the media stream?
I have a simple setup whereby I make my FWD account ring my Mediatrix 2102
as an extension to my Asterisk and the delay is horrific
If FWD is speaking IAX and the Mediatrix is SIP, * must remain in the loop,
even in theory.
If
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 converted
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
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 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 co
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 co
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 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 ano
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'
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 A
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 equipme
Hi,
Thanks for the replies.
Brian wrote:
FYI these so called "unlimited" monthly plans are RARELY,
if _EVER_ truly unlimited. They CAN (read the TOS), and
WILL terminate you if you use too many minutes more then
whatever average they calculated for when pricing the
plan.
I personally know several
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
se
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 tha
please take a look at these units:
http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1048701&CatId=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 some
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 p
y that I don't know why the suggested fixes didn't 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" <
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
http://www.ci
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 an
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 envi
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, Ethe
Hi Nik,
> Does shoutcast run across isdn?
Shoutcast, like its commercial counterparts, runs on an IP network.
You can:
1) use an ISDN modem (PCI, or PC card) that plugs into the PC, or
2) use an ISDN TA that connects via a serial port, or
3) use an ISDN router, connected via Ethernet to the PC.
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 oth
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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