similar to: incoming call label

On 02/15/2018 03:44 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 6:43 PM, thelma at sys-concept.com wrote: >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >> >> IN audocodes setting I have: >> "EndPoint Phone Number" >> >> Channel: 3 phone number: pstn-4444 >> Channel: 4 phone number: pstn-9998

On 02/15/2018 04:08 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:03 PM, thelma at sys-concept.com wrote: >> On 02/15/2018 03:44 PM, Joshua Colp wrote: >>> On Thu, Feb 15, 2018, at 6:43 PM, thelma at sys-concept.com wrote: >>>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >>>> >>>> IN audocodes setting I

On 02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, thelma at sys-concept.com wrote: > > <snip> > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn-4444 >> >> But asterisk display: >> Found peer 'pstn-9998' for

I've MP-114 that is working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension. Anything longer than 3-digits is cut off, example I dial extension 1000: [Apr 29 10:03:30] NOTICE[3817][C-000000e9]: chan_sip.c:25902 handle_request_invite: Call from '54' (10.0.0.115:5060) to extension

We have a caller ID from our phone provider "Shaw Cable" (digital phone) and it was working OK until recently. I get an error: WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have <4>, digest has <pstn-4444> NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user THELMA <sip:7804789998 at 10.10.0.103>;tag=50e17675d59121c4o1 at

I've Audiocodes MP-114 registered per-endpoint (2x FXO / 2x FXS) but when call comes on pstn-4444 it goes to context "fax-incoming" in sip.conf: [pstn-4444] type=friend context=incoming ... [pstn-9998] type=friend context=fax-incoming ... the device register per end point just fine, so it can find "secret=xxx" correctly but why the call is not forwarded to correct

Use the command bellow to check if is Asterisk opening the port. netstat -nap | grep 4569 You need to see something like this output, otherwise your asterisk is not opening the port. udp 0 0 0.0.0.0:4569 0.0.0.0:* 10244/asterisk Att, H?lvio Junior dCAA - Digium Certified Asterisk Administrator SafeId - Gest?o de identidades e Acessos +55 41 | 9

I think you need to increase verbose output and search in /var/log/asterisk/full for any error message related to IAX2 registration or simil. 2017-06-05 17:12 GMT-03:00 <thelma at sys-concept.com>: > No, I don't think it is IP table issue, I've not upgraded dd-wrt for a > while and it was zoiper was working OK with my previous version of > asterisk. > > After upgrade

You can use tcpdump in your server to verify if it is receiving the packets. tcpdump -ni any port 4569 So you have more than one ip in the server? On 5 Jun 2017 9:13 pm, <thelma at sys-concept.com> wrote: > No, I don't think it is IP table issue, I've not upgraded dd-wrt for a > while and it was zoiper was working OK with my previous version of > asterisk. > >

Another might be to make sure iptables isn't blocking the connection. You can run iptables -L -n -v To see if its set to block any ports. On June 5, 2017 9:06:55 AM EDT, thelma at sys-concept.com wrote: >I'm getting: >netstat -a |grep 4569 >udp 0 0 0.0.0.0:4569 0.0.0.0:* > >Should I be getting localhost IP? > >Thelma > >On 06/05/2017

Does asterisk listen on port 4569 by default? I'm running version Asterisk 11.25.1 and have a problem registering Zoiper (IAX) to Asterisk. I'm getting an error: Registration refused -- Thelma

I have two samba servers running Ubuntu 10.04 Samba Version 3.4.7 One server acts as domain controller and stores user ids in a .tdb Somehow I've ended up with a duplicate user name. On the Domain Controller # pdbedit -w -L|grep debbie debbie:1005:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX:84DEC6FE3B018B0FB977EDDF5009742C:[U ]:LCT-4D4B086F: On the other Server running winbind I get #

I'm using "insecure=invite" with two different dial plans, it it working with one dial plan but not with the other. What other parameters could influence "insecure=invite" In sip.conf below "insecure=invite" is working OK [pstn-1270] type=friend secret=spa3k username=voice-1270 mailbox=369 host=dynamic insecure=invite canreinvite=no disallow=all allow=ulaw

> On Jan 2, 2024, at 23:17, thelma at sys-concept.com wrote: > > On 1/2/24 15:13, asterisk at phreaknet.org wrote: >>> On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote: >>> I'm using asterisk-16.30.1 >>> >>> When I try to call another asterisk server over IAX I get a busy signal, >>> >>> chan_iax2.c:4739 __auto_congest:

The "error" I was talking about was in your log: "...== Spawn extension (extensions, 4, 3) exited non-zero on 'IAX2/home_server-6364'..." The call terminated here in a error which prevented the dialplan from continuing. Something there is broken, my recommendation is to check you registrations first inside asterisk: > sip show peers Something wasn't

I have two servers running Samba, one as a Domain Controller one as a Member Server. Both are running Ubuntu 8.10 and running smbd, nmbd and winbindd using the tdb back end. I am having a problem understanding ID mapping. The mapping is not the same on both machines. On the Domain Controller > root at thelma:/etc/init.d# wbinfo -i 'ATLANTA\rob' > rob:*:1000:2003:Robert

When I type: asterisk -rx "core show channels" I usually get Channel Location State Application(Data) SIP/pstn-4444-000003 7807574622 at internal: Up Dial(SIP/77807574622 at pstn-9998 SIP/pstn-9998-000003 (None) Up AppDial((Outgoing Line)) Is there a way to pull information about time the channel started? -- Joseph

Hi R users, I use plot.circular(rad, stack=T,bins=4) and could just obtain the four stacks in the directions of 45, 135, 225 and 315 degrees. But I want them in 0, 90, 180 and 270 degrees. Is there any parameter or any other function to make it? Thanks for helping a R beginner. -- Xiaohua Dai, Dr. -------------------------------------------------------------------------------- Centre for

On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote: > I'm using asterisk-16.30.1 > > When I try to call another asterisk server over IAX I get a busy signal, > > chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow > response >     -- IAX2/192.168.143.1:4569-656 is circuit-busy > > Asterisk-16.16 is working normally, no congestion error. There is not

I'm using asterisk-16.30.1 When I try to call another asterisk server over IAX I get a busy signal, chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response -- IAX2/192.168.143.1:4569-656 is circuit-busy Asterisk-16.16 is working normally, no congestion error. -- Thelma