Using cvs head downloaded as of just a few minutes ago..
chan_agent.c: In function `action_agents':
chan_agent.c:1446: warning: long int format, time_t arg (arg 7)
chan_agent.c: In function `__login_exec':
chan_agent.c:1684: syntax error before `char'
chan_agent.c:1701: `agent_goodbye'
gcc version 2.95.3 20010125 (prerelease, propolice)
on OpenBSD 3.6.
BJ Weschke wrote:
Compiled fine here. What version of GCC are you using?
On 11/3/05, Matt Hess [EMAIL PROTECTED] wrote:
Using cvs head downloaded as of just a few minutes ago..
chan_agent.c: In function `action_agents
We use tp-260 boards for ss7/sigtran.. they seem to behave similarly to
mp2000 or tp1610 series boards which we have used with both mgcp and sip
protocols.. their stuff seems to work rather well .. at least for us but
YMMV.
Chard Johnston wrote:
Hi All,
Does anyone have any experience with
zaptel.conf
Humberto
would you please share line 213 with us?
On 10/18/05, Matt Hess [EMAIL PROTECTED] wrote:
I have a customer that needs to do cas signaling across a t1,esf span..
it looks like this can be done but I'm not sure how as the documentation
is very light in regards to cas
Not that I've seen.. about all you can do is adjust the inter digit
timeout..
Louis-David Mitterrand wrote:
Hi,
I looked at the docs and probably missed it: is there a way to set a
dialplan on the GXP-2000? (to avoid having to press Send)
Thanks,
begin:vcard
fn:Matt Hess
n:Hess;Matt
I have this in sip show history for a particular channel marked as dead
(should be removed) in sip show channels:
1. TxReqRelINVITE / 102 INVITE
2. Rx SIP/2.0 / 102 INVITE
3. CancelDestroy
4. Rx SIP/2.0 / 102 INVITE
5. CancelDestroy
6. Unhold SIP/2.0
Attached is a pcap of sip packets that pertain to another call similar
to the history shown.. it's hard to nail these down as it takes a lot of
time, patience and sifting through dumps.
Olle E. Johansson wrote:
Matt Hess wrote:
I have this in sip show history for a particular channel
SIP/2.0 / 103 INVITE
17. CancelDestroy
18. Rx BYE / 201 BYE
19. TxResp SIP/2.0 / 201 BYE
And the packet capture is attached again..
Matt Hess wrote:
Attached is a pcap of sip packets that pertain to another call similar
to the history shown.. it's hard to nail
I have a customer that needs to do cas signaling across a t1,esf span..
it looks like this can be done but I'm not sure how as the documentation
is very light in regards to cas.. it would appear that I need to use sf
signaling but I get an error saying:
$ ztcfg -vv
Notice: Configuration file
I have been seeing the subject behavior on head for a few days now..
(been trying nightly builds to see if a bug causing this has been fixed)
on a sip show channels I get a little of active channels that I can
correlate calls to.. but I also have some dead channels listed that
should no longer
We use the mp-108 fxs units a lot.. also use mp-2000 units for pri_cpe
end. Probably the closest thing to your situation is our use of the
mp2000 terminating a pri at the z end and sending calls on to asterisk.
While it was not without it's flaws I can say that it worked rather well
just using
In light of the I/O bottleneck problem I'd have to ask why asterisk
can't just buffer incoming audio and then flush a complete audio file to
disk.. I'm assuming that recordings vary in length.. the problem with
this idea is what happens if 50 recordings all complete at the same
time.. a dump
I receive an invite from a vendor device..
U 2005/09/19 09:00:18.991139 66.185.96.32:5060 - 66.185.96.23:5060
INVITE sip:[EMAIL PROTECTED];userphone SIP/2.0..Via:
SIP/2.0/UDP voip.livewirenet.com:5060;branch=z9hG4bK6ee64f86
Max-Forwards: 70
To: 3036284320 sip:[EMAIL PROTECTED];tag=as60bbddbc
Just call a milliwatt..?
C F wrote:
Try your local DMV :)
On 8/18/05, Derek Whitten [EMAIL PROTECTED] wrote:
try calling comcast.. they are always good for at least 15 minutes of
hold 18778242288
qworst(qwest) works too.. 1800244
On Thu, 2005-08-18 at 06:28, Adam Vocks wrote:
Just
Just trying to get some resolution on this as to why it would work with
asterisk stable but not current.
*bump*
Matt Hess wrote:
I just updated one of my stable asterisk systems to head to test it
out.. and I'm receiving a interesting log message now in asterisk..
Aug 2 13:20:56 NOTICE
I'm looking for opinions on g726-32 vs. g711u..
They both have decent audio quality.. and looking at the wiki I get the
impression that g726 is like the little brother to g711. Yet, I've run
into quite a few sip termination vendors who don't support it. Does
anyone on the list actively use
Just checking to see if the list server died again.. been a few hours
since I sent something to the list and I usually see it by now..
begin:vcard
fn:Matt Hess
n:Hess;Matt
org:LiveWire Networks
adr;dom:;;4577 Pecos St;Denver;CO;80211
email;internet:[EMAIL PROTECTED]
title:Sr. Network Engineer
I just updated one of my stable asterisk systems to head to test it
out.. and I'm receiving a interesting log message now in asterisk..
Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce
received from 'sip:[EMAIL PROTECTED];user=phone'
(one line per registration)
I'm
Haven't seen email since the 29th.. just testing.
begin:vcard
fn:Matt Hess
n:Hess;Matt
org:LiveWire Networks
adr;dom:;;4577 Pecos St;Denver;CO;80211
email;internet:[EMAIL PROTECTED]
title:Sr. Network Engineer
tel;work:303-458-5667 x 106
tel;fax:303-458-5725
x-mozilla-html:FALSE
I should also expand this to note that I have also tested the amaflags
variable and it also has no effect on my sip peer entries' cdr records.
(latest stable asterisk has been tested)
Matt Hess wrote:
My cdrs are missing accountcodes for incoming calls from other
asterisk servers..
I've seen
We are receiving multiple audio drop outs on calls .. I've done quite a
bit of troubleshooting and it only involves calls that require the
Dial(SIP/xxx,,t) for transfers.. as long as the media path goes through
the server the audio blips happen.. using ulaw codec, btw. I have been
able to
Almost sounds like you and I have the same problem.. please check out my
message (I just sent to the list) and see if it rings any bells would you?
Stuart Lester wrote:
I am currently experiencing intermittent silences with my asterisk system.
The symptoms are as follows:
* Both for
Maybe I've missed it but I'm wondering if there has been any movement
towards getting t.38 support into asterisk.. has there been any news?
Where is t.38 support at? will it even happen?
begin:vcard
fn:Matt Hess
n:Hess;Matt
org:LiveWireNet
adr;dom:;;4577 Pecos St;Denver;CO;80211
Message-
From: Matt Hess [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 16, 2005 4:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] t.38 support news?
Maybe I've missed it but I'm wondering if there has been any movement
towards getting t.38 support
I, personally, think a channel driver handling both sigtran and
mgcp/megaco would be an ideal setup for bridging the gap between ip and
pstn.. especially with the current hardware devices on the market..
but all of that is just opinion..
Kevin P. Fleming wrote:
[EMAIL PROTECTED] wrote:
But I
Has anybody compiled app_conference as of late?
I've already asked on the app_conference devel list but as I'm rather in
a hurry my thinking is somebody here has both run into and found a way
to get this compiled and running.
Using stable asterisk and the most recent app_conference from it's
Just a thought I had on this.. Why not setup a sip peer entry in
sip.conf with a qualify statement in it and send the call to the peer
entry? That way asterisk will know if the peer is alive or not and I
would think it would skip that particular peer accordingly.. that or
it's really late and
I may be going out on a limb here but maybe not everyone is on both the
users and biz lists as not everyone wants commercial adverts in their email.
Note the name of the list.. Asterisk Users Mailing List -
Non-Commercial Discussion
sorta says it all right there..
Garrett Smith wrote:
Matt:
I
PROTECTED] On Behalf Of Matt Hess
Sent: Monday, January 03, 2005 7:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco Phones
I may be going out on a limb here but maybe not everyone is on both the
users and biz lists as not everyone wants commercial
I reported this on dev yesterday.. I thought I saw it fixed in dev but
not stable according to the cvs list..
Modified Files:
chan_sip.c
Log Message:
Minor ACk fix (bug #2687, again)
So the stable version is still borked.. but head should be cleared
up..heh, stable ain't that stable right
Fyi, just saw a fix go through for stable :)
John Hill wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Hess
Sent: Tuesday, December 21, 2004 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
is this current cvs or something? It looks completely abnormal for stable..
seems you are doing a lot of extra stuff you don't need to.. I'd see if
just this works for you..
exten = 800,1,Dial(SIP/800,60)
exten = 800,2,VoiceMail(800)
also.. why disallow all and then allow most everything? seems
system (loaded module).
-michael
On 12/2/04 1:07 AM, Matt Hess [EMAIL PROTECTED] wrote:
Does cvs tag v1-0 not have a dial command? I do not seem to have one..
dial
No such command 'dial' (type 'help' for help)
Henry Devito wrote:
Ok try this
Login into console
Set verbose 15
Dial
That's what I thought.. so that leaves me still at square one for
isolating this issue..
Andrew Kohlsmith wrote:
On December 2, 2004 12:12 pm, Matt Hess wrote:
Is an audio card a prerequisite for using voicemail? I never saw that it
was and such a requirement would seem absurd but I thought
Let me ask this.. does anyone have a suggestion for how I can track down
which file or directory asterisk is requesting to open?
Dec 1 12:17:11 WARNING[19092]: file.c:1058 ast_waitstream_full: Wait
failed (No such file or directory)
Matt Hess wrote:
That's what I thought.. so that leaves me
I'm having a problem with voicemail where the system will allow me to
login to the vm box no problem but when it starts tell tell me the
number of messages I have it hangs up.. I get you have and it dies
right there.. I'm running cvs tag v1-0.. what might be causing this?
I looked through my
into verbose mode and see what happens on the console
when you call it... that should help diagnose the problem.
On Dec 1, 2004, at 9:47 AM, Matt Hess wrote:
I'm having a problem with voicemail where the system will allow me to
login to the vm box no problem but when it starts tell tell me the
number
into verbose mode and see what happens on the console
when you call it... that should help diagnose the problem.
On Dec 1, 2004, at 9:47 AM, Matt Hess wrote:
I'm having a problem with voicemail where the system will allow me
to login to the vm box no problem but when it starts tell tell me
it went ahead and made a liar out of me and proceeded to not work the
second time around.. and really every time since then..
Matt Hess wrote:
I did some testing on this .. created a new vm box.. let vm record a
message to it's proper directory.. everything appears fine but again
PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Hess
Sent: Wednesday, December 01, 2004 11:47 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] voicemail cuts off / hangs up
I'm having a problem with voicemail where the system will allow me to
login to the vm box no problem but when
] On Behalf Of Matt Hess
Sent: Wednesday, December 01, 2004 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voicemail cuts off / hangs up
yup.. that's something I thought of as well.. and it's all there..
funny thing is.. I can start asterisk.. login just
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess
Sent: Wednesday, December 01, 2004 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voicemail cuts off / hangs up
Test completed successfully..
test dialplan:
exten = 555,1,Answer
Sorry for the OT message but I'm very curious to see if anyone on this
list uses pointone for long distance sip call termination?
We've been having an off and on problem with them saying they do not
support sip message with a fqdn in the from field.. which to me appears
to be a breakage of the
-0600, Matt Hess [EMAIL PROTECTED] wrote:
Remember, you pay for what you get.. especially with Dell networking
equipment. I have heard about several groups who tried the dell switches
only to give up on them because the dell switches just didn't perform.
Yes, price-wise they look good
I haven't found much info that stands out to me in regards to echo
cancellation in sip to sip calls..
My setup is this..
I have a key system connection to an audiocodes mp-108 which connects to
asterisk via sip and asterisk passes the call over to a ser proxy and
ser passes the call to either
A hears echo and
point B does not then it's something like an impedance problem on the
far end.. or wire center (CO) at that point.. but an impedance mismatch
can occur also at a pstn ingress point such as the ILEC side of the pri
lines.. or maybe I'm wrong..
Steve Clark wrote:
Matt Hess wrote
I use a failure route in ser for the call to be sent to the voicemail
system..
I use ser as mainly a primary router for sip messages that sits in the
center of a ring of asterisk servers that feed the clients..
Iqbal wrote:
Hi
i am stuck with the same dilemma, as the original poster
I have
Ironically, I just got back to my desk from faxing an 18 page document..
ulaw codec..
call path: sip gateway - asterisk - max tnt - pri lines/pstn
I haven't really had many troubles with fax.. apart from when I was
using a particular voip provider named voiplist.. they refused to
support it
Remember, you pay for what you get.. especially with Dell networking
equipment. I have heard about several groups who tried the dell switches
only to give up on them because the dell switches just didn't perform.
Yes, price-wise they look good.. but as far as performance goes.. (that
is
Is there any option in Asterisk to create a quiet termination? I'm
looking for something similar to the 600 ohm impedance lines that COs
used to have to check for return loss.
begin:vcard
fn:Matt Hess
n:Hess;Matt
org:LiveWireNet
adr;dom:;;4577 Pecos St;Denver;CO;80211
email;internet:[EMAIL
to work (so far).
Steven Critchfield wrote:
On Wed, 2004-10-13 at 11:26 -0600, Matt Hess wrote:
Is there any option in Asterisk to create a quiet termination? I'm
looking for something similar to the 600 ohm impedance lines that COs
used to have to check for return loss.
wouldn't that be like
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