similar to: Deutsche Telekom: calls dropped after 15 minutes

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Hi list! Very strange... I ran the Asterisk CLI for other tasks, and suddenly I got this message: == Using SIP RTP CoS mark 5 -- Executing [000972592603325 at default:1] Verbose("SIP/192.168.20.120-0000002a", "2,PROXY Call from 0123456 to 000972592603325") in new stack == PROXY Call from 0123456 to 000972592603325 -- Executing [000972592603325 at default:2]

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer

> Very strange... > I ran the Asterisk CLI for other tasks, and suddenly I got this message: > > == Using SIP RTP CoS mark 5 > -- Executing [000972592603325 at default:1] Verbose("SIP/192.168. > 20.120-0000002a", "2,PROXY Call from 0123456 to 000972592603325") innew stack > == PROXY Call from 0123456 to 000972592603325 > -- Executing

For such cases i created a dialplan in the default dialplan which blocks the ip of the hacker with iptables. On Monday, June 8, 2015, Luca Bertoncello <lucabert at lucabert.de> wrote: > Hi list! > > Very strange... > I ran the Asterisk CLI for other tasks, and suddenly I got this message: > > == Using SIP RTP CoS mark 5 > -- Executing [000972592603325 at

I'm guessing this is a small/home system? I suggest you install SecAst from this site: www.telium.ca It's free for small office / home office and will deal with these types of attacks and more. It can also block users based on their Geographic location (based on the phone number it attempted to dial I suspect this is middle east), look for suspicious dialing patterns, etc. If you

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > Make sure you have solved the problem. You don't want to get hit with a > phone bill for calls from your location to Israel. Basically, they are > hoping that you are running the equivalent of a mail server open relay. > They are trying to use you to dial out to another number. You don't want > to pay

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

> I think there are many german users in this ML, that use Asterisk with the > new line of Deutsche Telekom (Magenta Zuhause). > > My ISDN will be converted in Juli (Kaboom-day at Juli, the 3rd...) and right > now I can just hope, that I configured my Asterisk well to work with Deutsche > Telekom, but I cannot be sure, since I can't test it... > > So my question: can

Am 13.06.2015 um 13:54 schrieb Luca Bertoncello: > I think there are many german users in this ML, that use Asterisk with the > new line of Deutsche Telekom (Magenta Zuhause). I don't think so. Most users will use the router provided by Telekom. Anyway, after 15 seconds of Google'ing for Magenta Zuhause and SIP, maybe this will help you:

Zitat von Sebastian Kemper <sebastian_ml at gmx.net>: Hi Sebastian > Brian suggests to check the SIP traces. You can either enable SIP > debugging in Asterisk like so: > > sip set debug on > > Or you could run tcpdump and capture the SIP traffic. > > The first option is probably the easiest. I tried with sip set debug 42 sip set verbose 42 The result was

Zitat von Sebastian Kemper <sebastian_ml at gmx.net>: > I don't remember seeing anything looking like a SIP trace in your first > mail. Try > > sip set debug on > > instead of > > sip set debug 42 > > I don't think there's a sip debugging level like 42 in Asterisk. You can > either switch it on or off. Is it not this:

jg <webaccounts173 at jgoettgens.de> schrieb: > It doesn't really depend on your sip.conf and Asterisk. Your gateway/router > will be the major problem. My summer project will be to look at session Are you sure? Right now I'm using an italian SIP-Provider (Messagenet), configured in my sip.conf and I can receive calls without any problem... So, I don't think, I have to

Am 23.06.2020 16:22, schrieb Marek Greško: > It seems your problems lie in something other. Most probably it is not > mtu problem. All my suspections are contradicted. If it is true you > have inter vlan voice quality problems, it is definitely something > different. Formerly I assumed you were trying only LTE vs LAN using > internet. I'm not sure what you mean with the last

>> It doesn't really depend on your sip.conf and Asterisk. Your gateway/router >> will be the major problem. My summer project will be to look at session > Are you sure? > Right now I'm using an italian SIP-Provider (Messagenet), configured in my > sip.conf and I can receive calls without any problem... > So, I don't think, I have to expect problem on my NAT