I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet,
and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet
parameters are all set correctly in sip.conf. An inbound call from Sipphone
works great until the local channel places the call on hold. During
I've scoured the web for hints, and find a lot of chatter about one-way
audio with IP Kall, but no definitive explanation. I have the default range
(5000-31000) of UDP RTP ports forwarded right to my asterisk box, and have
no other difficulties with one-way audio on any other peers. Does anyone
SPA504G - 1 more vote for it.
It is worth having 4 lines even if you need 1 initially.
SPA504G supports G722 and sound is awesome even if you do not not use
teh HD sound. If you do not care that mcuh about HD sound and do not
need PoE SPA941 is a excellent choice - you get really a lot for the
Set the primary unit to pick up on, say, 2 rings and the secondary unit
to pick up on 4.
If the primary fails, it won't pick up the line and the secondary takes
over.
You could try having your phones register to both but I'm not sure how
that would work
regards,
Drew
That's exactly how I
List,
A Cisco 7960 is registered to servers A and B, where B is the backup server,
only used by the 7960 if A is unreachable. That is the behavior of these
phones.
A call comes from server B to the 7960, which is successful. The 7960 then
tries to park the call via an attended transfer, so the
Does anyone have a solution for parking calls to the most local asterisk
server using SIP?
Situation: incoming zap call to server A is answered by a phone registered
to server B, but when parking the call, the phone parks the call at server B
instead of server A, thus introducing unnecessary
List,
Question about the Polycom 650: when dialing the digits for a phone number,
and an incoming call comes in, does the phone prevent you from completing
your outgoing call until the phone stops ringing, like a Cisco 79X0 does?
--Brent
___
--
Upgraded to 1.4.17 and found that the parking slot is not announced.
Reverted back and all is well. Anyone else notice this behavior?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
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Does anyone have any tricks to allow codecs based on what network a phone is
on? i.e., allow uLaw if the device is on the LAN, and only allow g729 if
the device is anywhere else?
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
tel:+1 219 836
Running 1.4.11, and during an established SIP call, we often get audio drop
outs if another call comes in. Anyone else see this happening? Incoming
calls ring both some local SIP phones, and also some other servers via IAX
trunks.
--Brent
___
Sign
Does anyone have any tricks to use some logic with SIP UA's codec
negotiation based on the UA's IP? What I would like to do is have Cisco
7960's use g711u when they register with a local IP, and g729 when they
register with a non-local IP. I was thinking about sip.conf and making two
entries for
Hola,
What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXX?
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:[EMAIL PROTECTED]
web:www.torrenga.com
When trasferring a call, how is the context determined?
When using a zap device, and the DTMF code for blind or attended transfer is
entered, does the tranfer originate at the context the zap device is set to
be in, or does it originate from where the outside call being transferred
originated in,
I discussed this with Digium techs, who recommended using the 1.4 version of
Zaptel with the 1.2 version of Asterisk, at least with my hardware
(TDM400P). The 1.4 zaptel does not yet support the HPEC (which won't run on
my system anyways...), but does have a totally rewritten MG2 EC. I am
running
List,
Has anyone got any hints of how to setup the OpenSuSE Firewall2 with VOIP
friendly traffic shaping? The only bit I see is in the config file regarding
how to setup a simple HTB. I come from Shorewall, and am finding this
firewall to be different. Any help is appreciated.
Sincerely,
Brent
I am thinking of turning on the NAT option in our Cisco phones (and the
corresponding sip.conf modification) to allow the phones to be taken outside
the LAN.
Can anyone think of any reason not to just always turn on the NAT enabled
option? I can't think of a reason not to always operate these
Dan,
Try a cron job to place a call file into /var/spool/asterisk/outgoing (I
think, sorry, don't have access to that machine right now).
I do a similar thing with wakeup calls - a cron job puts a call file into
the above dir that rings my phone each weekday morning. There's no reason
you
List,
Sometimes, when doing an ENUM on US toll-free numbers, we get the following
response when dialing to tf.voipmich.com:
Forbidden - wrong password on authentication for INVITE to 'Torrenga
Engineering sip:[EMAIL PROTECTED]
Can anyone explain what this means, or how to avoid it? It does not
Yeah, I thought dnsmasq was the cure, too. We had an internet outage last
week. It was odd, our ISP (ATT) changed out static IP's (don't ask... No
one knows why... At least I figured out what was going on...). Thus, our
modem/router was whacked, as well as our firewall. So, I think every piece
of
Anyone notice that tf.voipmich.com (ENUM for US toll free service) will
connect you successfully, but then disconnect after what seems like 30
seconds or so? Anyone know what might be going on here? I googled the hell
out of voipmich and did not get very far.
Sincerely,
Brent A. Torrenga
Is there anyway to force an autopickup on a cisco 7940 / 60 from the
dialplan ?
Unfortunately, no.
However, the 7940 can use two SIP accounts, the 7960 six accounts. This is
how I implemented intercoms:
Set the first (top) line to register to an account that will be used to
ring, set the
I think [EMAIL PROTECTED] allows a user to search a directory by either first
OR last
name, right? I don't know for sure since I don't run [EMAIL PROTECTED]
I would like to offer that functionality in my system - and I'd have done it
by now if there was a prompt where Allison asks press 1 to
|general|ext-local|bo) in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/directory
-- Playing 'dir-intro-fnln-oper' (language 'en')
On 6/14/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Brent Torrenga wrote:
I think [EMAIL PROTECTED] allows a user to search a directory
Dear list (and more specifically Bret),
I am getting one-way (inbound only) audio when trying to place a SIP call
via voip.trxtel.com (i.e. [EMAIL PROTECTED]). The Cli spits out
== Forcing Marker bit, because SSRC has changed 5 times after atempting a
native bridge. I realize this is most
If you need to do a couple differing operations on a list of many
area/country codes, then you may consider using the database to let the dial
plan choose what to do, rather than go through so many extensions.
I mention this to keep your extensions.conf easier to read, not because I
know whether
Might want to take a look at http://web.csma.biz/apps/xml_xmldir.php for a
starting point.
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:[EMAIL PROTECTED]
web:www.torrenga.com
Is it possible to dial more than one extension with a different CID to each
extension? I'm thinking macros might be needed, but I don't have a good
handle on macros. Is it possible? Any hints?
BTW - this would be used for showing an internal extension to one phone and
a PSTN accessible number to
Doug,
I think I have seen this - does the initial invite to pbx2 make complete
sense - is it valid according to the sip.conf entry on pbx2? In the pbx2
sip.conf, try insecure=invite, maybe insecure=invite,port/insecure=port. I
don't have a real handle on why, but I recall it solving some sort of
Hello,
Anyone try to use SIP TAPI (http://www.enum.at/index.php?id=479) with
Asterisk?
Pretty nice, pretty simple. I am hung up on something, though, and google
doesn't specifically address my issue.
The program seems to go to the s extension in the default context of the sip
user it is
Clint,
Crap. Wish I would have seen your setup first. I played with asttapi for a
few days, and gave up. My problems were manager related, and you cover those
points well enough on your page.
I was able to get SIP TAPI to work this way:
- each install of SIP TAPI needs a SIP user in sip.conf.
So I saw mention of a way to allow dialing using SIP URI's on Dave McNett's
site at http://slacker.com/~nugget/projects/asterisk/page7
Wow, awesome, I can call anywhere now. However, I think there is a more
elegant way of figuring out whether or not the local * server should handle
a given
Howdy,
How can you tell if RTP traffic has been reinvited/is bypassing an * server?
Sincerely,
Brent A. Torrenga
[EMAIL PROTECTED]
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
+1 219 836 8918 x325 Voice
+1 219 836 1138 Facsimile
www.torrenga.com
List,
Per someone's suggestion (thanks, whoever you were) from this list, I
implemented dnsmasq to prevent the issue of resolving DNS when the net
connection goes down.
This morning the net connection was down, and our * server didn't miss a
beat. I recommend looking into:
Dearest List,
I understand how to handle guest calls via SIP and IAX. However, when such
a call is placed, it will not look like IAX/guest-1234, or SIP/guest-1234.
Instead, it will be something like IAX/the.callers.ip.address-1234
My issue is with getting this to map to a Flash Operator Panel
Ken,
Try using the CALLERID function, instead of setting the variable.
As in: exten = _XXX,2,Set(CALLERID(number)=6031234${CALLERIDNUM:1})
Or maybe you need the greater/less than 6031234${CALLERIDNUM:1} dealies.
See: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid
Hi,
I have got my setup almost how I would like it now, but I have just
two last remaining issues that I cant seem to find answers too so i'd
be grateful if someone could help?
1) Since upgrading my Cisco 7960 SIP phone to P0S3-08-2-00 the phone
now displays the IP address of my asterisk server
Thanks for letting me know regarding the @ip-address problem. I take
it you have experienced something similar with this firmware?
Yup. Due to this I only run 8.2 on my phone, and use 7.4 on any other. You
can roll back to 7.4 and be ok, don't use 7.5 as it has a bug relating to
the phone
It doesn't seem as much broken as just annoying. I am holding off on
upgrading until this resolves, but it doesn't seem to affect performance,
anyways. BTW, some folks say that the server address only gets appended to
the CID when a redirect or something comes about. Our experience here shows
that
What would cause this? It happened out of the blue:
-- Executing VoiceMail(Zap/3-1, [EMAIL PROTECTED]) in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/6' (language 'en')
Anyone experience the double ringing when calling out over TelIAX? I am
using a Cisco 79[46]0, and do not use the r option in the Dial() command.
I always thought that the r is what causes double ring, and is never
really needed except to cause problems...
Sincerely,
Brent A. Torrenga
[EMAIL
Has anyone else noticed that notifying a 79[46]0 with cisco-check-cfg
doesn't elicit any response from the phone using fw 8.2?
Sincerely,
Brent A. Torrenga
[EMAIL PROTECTED]
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
+1 219 836 8918 x325 Voice
+1 219 836 1138
Out internet connection was out this morning. It seems that the SIP
extensions on our LAN were affected. Behavior like:
Call comes in over POTS to a TDM400P, there is a delay then before the Cisco
79[46]0's start to ring.
If we were lucky enough to get a call through, then we could not transfer
Our user places a call, the gateway responds with no sound at all, or
hangs up, or gives busy tone.
How can we get to the next provider?
I have now:
exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
;exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
;exten =
Brent,
you mean, I could just remove the remark signs and number it 103, 104,
105, since it does not matter why it failed (busy, congestions)
(maybe for statistic purpose to add a log entry for the move to the next
provider).
bye
Ronald
Yup. Take a look at the macro solution, too. I
We use them for origination over IAX. At first we had callee's reporting
that our voice was choppy to them, while the callee has always sounded fine
on our end. I made that problem go away by introducing traffic shaping at
our firewall.
They have a bug, whereby on their website you can set the
Forgoe the patch, just upgrade to 1.2.6. The changelog lists it as a fix
from 1.2.5 to 1.2.6.
I'm a bit newbie, could you tell me how to i apply the patch?
Thanks in advance
Marco Mouta
On 3/27/06, Benoit Panizzon [EMAIL PROTECTED] wrote:
On Friday 24 March 2006 16:05, Benoit Panizzon wrote:
Dollars to donuts it is related to these two posts, but no one seems to know
where or why it happens - this issue doesn't seem to be related to one
specific piece of hardware:
Post 1)
*
Anyone ever seen MeetMe cause * to crash?
Anyone ever seen MeetMe cause * to crash? Specifically, it happens
consistantly if someone begins to enter a conference and then decides to
hangup while Allison is introducing them - like playing back
conf-onlyperson. This has been seen with the MeetMe participant connecting
via IAX and SIP (not
Paul,
Ah, I see. Our echo is largly under control now. It took me a while to
figure out the gains and get them tuned, and now the echo only leaves very
small artifacts. Nonetheless, this still provokes the odd complaint here and
there. We use VOIP for outgoing calls when our POTS lines are
Howdy:
Does echo only occur on analogue PSTN lines, or can it also occur on PRI and
BRI lines? If so, for the same reasons? This is a part of our consideration
to transition to BRI.
Sincerely,
Brent A. Torrenga
[EMAIL PROTECTED]
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana
Try 7.4 (P0S3-07-4-44). Notice on voip-info.org the comments about 7.5 being
buggy with some sort of registration issues (if I recall). Also, voip-info
reports 7.4 to be stable. If you read the Cisco changelog, you'll see LOTS
of bug fixes up to 7.4 (no new features, just fixes). I THINK one of
chan_zap.so make all changes
effective, or do I need to reload or service asterisk restart?
--Brent
Brent Torrenga wrote:
I have one use on our PBX who has been experiencing seemingly random
disconnects. The user is on the same LAN as everyone else, using the same
Brent,
The last time I was having
FOR THE LIST'S BENEFIT, THIS IS MY EMAIL TO THE LOUD PARTY ON OUR SYSTEM,
THANKS FOR ALL YOUR HELP, HOPEFULLY I HAVE THE ISSUE SOLVED:
Well, I got a series of suggestions as to how to solve your hangup problem.
My favorite suggestion:
LOL... You could try
Andrew: Yeah, busydetect=yes == problems. Duly noted!
Alexander:
Sorry, not a PRI line, just a TDM400P.
Would a PRI or BRI not use the D channel to signal busy, anyways? I have a
lot to learn about the workings of ISDN...
One other thing that I did not mention, Are you using a PRI? What are
Well, I thought and hoped my issue of random hangups on our TDM400P were
related to busydetect=yes in zapata.conf. The behavior of a call being
hungup has not changed, however, since setting the busydetect option to
'no'. Again, the only affected user is my loud talker...
What are some
I have one use on our PBX who has been experiencing seemingly random
disconnects. The user is on the same LAN as everyone else, using the same
type of phone (79XX loaded with SIP firmware) as everyone else. He had some
disconnects a few weeks ago, I suspected the phone, so I swapped his with
mine.
We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This
should get rid of static on the line (at least any static generated by our
half of the circuit), right?
I am a total virgin to ISDN. I understand that we need two BRI circuits to
provide four voice channels, and that the
The native MOH type will, unless set to random=yes, play the music files in
the same order as they appear with an ls of the directory. (someone, anyone,
back me up here?)
I would place the greeting in the same MOH class as your actual music, and
name the file of the greeting something less than
... Ymmv
--Brent
In theory that is fine, however, when a call leaves the MoH, the MoH
doesn't
stop playing. So when a second call is received, the MoH is still playing
and therefore they dont hear the message for a few minutes.
Dan
On 02/02/06, Brent Torrenga [EMAIL PROTECTED] wrote:
The native
Thanks for your input, everyone, but I still think it is on Teliax's end...
I will present our collective thoughts to their tech.
Kevin,
I am using IAX. When I turn on IAX debug, I get:
--SNIP CLI OUTPUT--
-- Executing Dial(SIP/Brent_ring-bcf7, IAX2/teliax/18005558355) in
new
stack
Might it be related to the memory leak bug? Upgrade to 1.2.4? (shot in the
dark, a brainstorm on my part is all)
Here's what the logfile shows. Any ideas? And is
there a way to fix the deadlock without restarting Asterisk?
Feb 1 09:10:33 WARNING[5327]: channel.c:784 channel_find_locked:
Has anyone had problems getting their preffered codecs on the Teliax web
interface taking effect?
I have two accounts, two separate yet similarly configured * servers. On one
account the settings took right away - on another server I am getting no
result. In fact, no matter what I change the
Dearest List,
I guess I missed this point: Is it true that if you change the echo canceler
in zconfig.h, and then recompile/install your zap modules, that for this to
be taken into effect by * you must then recompile/install *?
I would have figured that the zap echo cancellation method was
Another piece to the puzzle, for what it's worth:
The last moments before the crash, an incoming Zap call was answered by a
SIP phone, parked, and then picked up by another SIP phone. During the
picked up conversation, the audio was reported to me to be patchy, described
as cell phone like. It is
It looks like my modules are all up to date - they are all dated the 25th
Jan - aka Black Wednesday.
What really bugs me about this is the lack of useful info from any logs. The
last call to take place, the call that gets distorted, has no entry. This
has to indicate something, no?
Hmm - I'd do
Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
hours or so.
Since upgrading to 1.2.3, though, the whole system has locked up twice. Once
on Thursday, and then about a half hour ago. The server
Yeah, got two whacked 1.2.2 servers here. Updated to 1.2.3 from ftp source,
and a-ok.
HOWEVER, echo cancellation seems to be non-existant in our TDM card?!? I
recompiled/installed zaptel 1.2.2, no effect. Even tried different
cancellers. Anyone else experience this after upgrading to 1.2.3?
Well, I want to divert an incoming fax call to just play an error message
(please hang up, call my fax number...), rather than drive my secretaries
nuts. I have zapata.conf as faxdetect=incoming, and, in fact, I will get the
notice at the CLI that Fax detected, but no fax extension. I guess that
I think he is getting at something like a Zap channel that passes on it's
own CID info from zapata.conf, as opposed to the calling channel? Perhaps it
is a zap issue, and is as simple as placing callerid=asreceived in
zapata.conf.
OR
Maybe it is the way Dial() works in 1.2 versus 1.0 - with the
I use Cisco 7940's and 7960's over here, loaded with SIP, and they do very
nicely. Also, the SCCP channel for * is under heavy development, and may
offer a future option to convert in that direction, too (SCCP, or skinny, is
their native tongue, not SIP). We got our phones from John Putnam at
I use Cisco 7940's and 7960's over here, loaded with SIP, and they do very
nicely. Also, the SCCP channel for * is under heavy development, and may
offer a future option to convert in that direction, too (SCCP, or skinny, is
their native tongue, not SIP). We got our phones from John Putnam at
Check out the Cisco SIP IP Phone Administrator Guide, Appendix D -
speed_line and speed_label
Do a google for Cisco SIP IP Phone Administrator Guide, easy peasy nice n
easy.
Sorry about the unrelated questions about cisco phones, but does anyone
know how to set the second line up as a
speed
Can someone explain how to use groups? I can't seem to wrap myself around
this, though I know it is something simple.
I have 3 zap lines, and when placing an outgoing call, would like to 1) use
a zap line if and only if 1 or fewer zap lines are being used at the time,
and 2) if more than 1 zap
I use IP Kall to forward my missed cell phone calls to. This way, if my
phone is off, or out of a service area, calls will go to my * box.
Concurrently, all incoming calls to my * box cause it to dial my local
extensions at home, my extension at work, and my cell phone via NuFone.
Problem: A loop
When dialing in after hours callers get to use the directory. I know
that if you put h or H with a Dial() command you get the behavior
of being able to terminate a call by pressing *. However, nowhere in
the entire extensions.conf does there appear the h or H option, so
I know it is not
I have my dialplan setup the same, only with 0 instead of * as the
extension. What would the reason be, after authenticating, that I get a
dialtone, as expected, but no response to any DTMF tones I input? It is as
if the DISA works, gives me tone, but is unresponsive? The destination
context is
So within the /var/lib/sounds/voicemail structure are the greeting files
recorded by the person at each extension (busy.wav, greet.wav). If I need to
get rid of the customized recording, it is trivial to simply delete both of
those files. At that point, if a call goes to voicemail, then Allison
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