Displaying 20 results from an estimated 1300 matches similar to: "Asterisk Realtime Unregister"
2010 Mar 12
4
Can not enable sip debug because CLI flooded
Hello list,
I have nat=no and qualify=no in my sip peer definition and still my CLI
is flooded with :
[Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985
handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms /
2000ms)
[Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985
handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (24ms /
2000ms)
[Mar 12 10:17:26]
2005 Aug 22
1
Qualify time +2000ms?
Although I'm convinced that Broadvoice doesn't have the most stable of
ping times, it seems like I get ping results that are approximately the
ping time +2000ms at times. Has anyone experienced this problem with
qualify on a SIP connection before?
So here, was the ping 20ms or 2020ms as reported?
Aug 22 06:39:49 NOTICE[6964]: chan_sip.c:8481 handle_response_peerpoke:
Peer
2008 Sep 22
1
I can't call my remote users?
Good day to all--
First off let me say that I have been very pleased with the mailing
list. I have learned a ton of stuff just reading other peoples
questions and comments. I really enjoyed the VOIP Conference call on
Friday morning. Still working on figuring out the best approach to
custom voicemail emails (the reason I joined this group); however, we
have more pressing issues. I
2008 Oct 07
1
regcontext
hi all,
just wondering what's happening here:
i have a pap2 and an spa941. everytime i call my spa from my pap2 i can
see it being added dynamically on the regcontext:
[Oct 7 11:59:08] -- Saved useragent "Linksys/SPA942-5.2.8" for peer
100100
[Oct 7 11:59:08] -- Added extension '100100' priority 1 to
sipregcontext
but from spa to pap2 i dont see it, i looked
2011 Jun 07
1
tls/srtp: sip_xmit error: returned -2
I'm having trouble setting up tls/srtp secure communications on my
Asterisk server- I'm still rather new at working with Asterisk.
I have enabled tls and encryption and I have csipsimple with tls build
on the phone. I'm currently only testing one phone with this capability
so far, and the rest still work in the current state.
My logging looks like this with verbose turned up:
2008 Jul 01
3
music on hold realtime
Hi,
Is it possible to use realtime for Music On Hold?
Is it also possible to store the music/audio files on the database, same
way a voicemail can be stored on the database?
Thank You
Regards,
Nhadie
2007 Aug 09
2
Asterisk Help
Asterisk Users,
I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service.
I have two Netgear switches on my T1 router, one for VOIP and another for
data.
I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for
all data. This morning I saw this message a few times on the Asterisk
command line. The lagged cause garbled phone calls.
Is my network to
2009 Apr 07
1
i have a probleme and my asterisk and ovh
hello every body
my connexion on ovh to pass in UNREACHABLE and not reidentified were not
reboot the server.
[Apr 7 20:17:21] NOTICE[19947]: chan_sip.c:15605
handle_response_peerpoke: Peer 'ovh' is now Lagged. (2067ms / 2000ms)
[Apr 7 20:17:35] NOTICE[19947]: chan_sip.c:19829 sip_poke_noanswer:
Peer 'ovh' is now UNREACHABLE! Last qualify: 2067
but my probleme is the adress
2008 Aug 22
4
set callerid with plus sign
Hi,
Is it possible to assign a plus sign on the callerid(num) ?
currently this is what i do CALLERID(num)=+6523450017
but telco is denying calls, coz they said they are seeing "bs523450017"
instead of +6523450017.
i tried putting it inside double quotes CALLERID(num)="+6523450017"
telco says the same thing.
is this possible? thank you
Regards,
nhadie
2009 Apr 26
1
Error, Clue to what?
[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer
'3516533812' is now UNREACHABLE! Last qualify: 86
[Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke:
Peer '3516533812' is now Reachable. (98ms / 2000ms)
[Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 -
2008 Apr 23
2
prepaid on the trunks
if i have this setup:
[sip users] -- [asterisk] --- [as5300] --- [pstn]
asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the asterisk so sip users can call out to pstn.
what i would like to is do prepaid on those trunks, not on the sip users. sip users can call any other sip users . i want to do it that way coz i'm trying to build a multi-tenant pbx, and i will use
2008 Jul 22
1
issue with high latency
Hi,
Is there a specific latency that asterisk accepts? I encountered a
problem wherein when the latency was unusually high,my xlite's (i have 2
xlite) cannot register. but when the link suddenly went stable, the
x-lite just registered. what i forgot to look at is if the registration
packet is reaching my asterisks.
------ when xlite cannot register ---------------
Pinging
2009 Jan 28
4
route based from source
Hi,
Is it possible to detect where the call came from and route it out to
different sip trunks.
e.g.
i have user 100300 when that user calls outbound i will make him use of
[sip-trunk-100]
another user, 101300 when that users calls outbound i will make him use
of [sip-trunk-101]
actually the 100 and 101 at the beginning of the username is the
accountcode i used for cdr.
hope my question
2008 Feb 27
1
SPA3102 registration problem
Hi list,
After failing to get a Sipura/Linksys SPA3000, which I've configured
as a PSTN gateway, to pass on the Caller ID, I decided to try my luck
with a Linksys SPA3102 after hearing some promising stories.
Unfortunately, I've run into a completely new problem: it seems
Asterisk won't let this device register.
I went about configuring the SPA3102 in much the same way as I
2009 Feb 18
3
US DID
Hi,
Anyone knows a DID provider that can do both outbound and inbound?
Regards
Nhadie
2010 Nov 24
2
SPA942 on speaker phone does not hang up?
Hello all,
I am using Linksys SPA942 in my current installation activity. I see a
peculiar behavior: A call is made and the SPA942 uses its speaker. When the
far end of a call hangs up , the SPA942 stays off hook, and after a time
plays a fast busy. The user then has to press the line presence button to
hang up the phone.
I think I must be missing some sip.conf parameter. My sip.conf is pretty
2006 Oct 15
2
SPA942 quality for a Bank
Before committing to about 50 of the spa942's, I like to take a last
poll from those on the list to identify any negative issues that might
be associated with the audio, functionality, early failures, etc, on the
spa942.
Expecting to deploy these using existing cat5 cabling and both rj45
jacks. Been using three of theme in a short term demo with the customer,
but the demo systems has
2008 Jun 11
2
time on asterisk
Hi,
I'm using gotoiftime on asterisk, but it seems there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob?
Regards,
nhadie
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2008 Oct 22
3
asterisk video
hi,
hs anyone able to make video to work on asterisk? i tried following this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam
i can see that eyebeam is trying to broadcast a video but the other
eyebeam is not receiving it.
i tested the same setup but this time using ser with rtpproxy and
eyebeam video works fine.
any ideas? where do you think should i start
2007 Jan 11
2
calls to SPA942 disconnect after 15 seconds (chan_sip.c set_destination: can't find address)
Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem.
For this device (when receiving calls) I get periodic "chan_sip.c set_destination: can't find address for host"
I have set the "canreinvite=no" in the sip.conf. Does anyone have a sample entry from sip.conf for the Lynksys SPA 942