similar to: Calling from "extern"

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: >

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 =>

-----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! 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

> 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]:

I'd start by turning on sip debugging in asterisk >sip set debug ip [your_phone_ip] and use tcpdump or wireshark to see what the OS sees tcpdump host [your_phone_ip] and udp port 5060 On 15-05-28 03:58 PM, Luca Bertoncello wrote: > 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

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,

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

Am 13.06.2020 um 08:28 schrieb Luca Bertoncello: > Hi! > > I have a Asterisk installation to manage my phones at home (provider is > Deutsche Telekom). > It works, but very often the voice is "broken"... > Yesterday during a call it was very difficult to understand what my > partner sayd... > > It can NOT be a problem of other downloads/uploads, since in that

Hi list! My Problem: all calls to international numbers will be dropped after exactly 15 minutes... I have a VoIP-account by Deutsche Telekom. This is what I see when I call someone (my parents) and the connection will be dropped: == Using SIP RTP CoS mark 5 -- Executing [+39015222222 at default:1] Set("SIP/00493511111111-00000125", "newNumber=0039015222222") in new

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

Darryl Moore <darryl at moores.ca> schrieb: > I'd start by turning on sip debugging in asterisk > >sip set debug ip [your_phone_ip] Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172.16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP

Hi list! So, new day, new problem... I tried right now to call from my cellphone a peer in my Asterisk. The cellphone has correct credentials, but it's NOT registered on my Asterisk, now. I just tried to call a peer in my network, from a peer not yet registered. And it works... :( The very curious thing is, that I can't find how the call will be accepted... Every section in my dialplan

Hi list! I'm using Asterisk 13.14.1 from Debian 9 repositories. The provider is Deutsche Telekom und Messagenet (just for receive). I can call and receive calls, but I have a little problem: there is a "delay" of about 1-1,5 seconds between the time the voice is sent and the time when the voice is received, so that it happens very often that the peer does not get my voice and try

Subject says it all. How can I get the 1.3 version and 0.9.28 to compile on CentOS 5.8 ??? When I compile the two as modules I get errors. My Makefile is: obj-m += 8139cp.o 8139too.o all: make -C /lib/modules/$(shell uname -r)/build M=$(PWD) modules clean: make -C /lib/modules/$(shell uname -r)/build M=$(PWD) clean The errors I get are: Entering directory

> Darryl Moore <darryl at moores.ca> schrieb: > > > I'd start by turning on sip debugging in asterisk > > >sip set debug ip [your_phone_ip] > > Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172. > 16.34.133' Method: OPTIONS > Reliably Transmitting (no NAT) to 192.168.200.11:5060: > OPTIONS sip:00493512222222 at

CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/../voicebus.o LD [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/wcte12xp.o

Hi all, I need to compile alsa-project 1.0.16 on the latest centos 5.1 kernel. I am getting this error. What to do... ? CC [M] /home/silentm/MessageNet/alsa-project/alsa-driver-1.0.16/acore/sound_oss.o CC [M] /home/silentm/MessageNet/alsa-project/alsa-driver-1.0.16/acore/info_oss.o In file included from

Hi Luca Y need to check your wifes codec priority list -seems to be GSM on the first place. Luca Bertoncello <lucabert at lucabert.de> wrote: >Hi list! > >I noticed that when the phone of my wife calls the gsm codec will be used, >but if someone calls the phone, alaw will be used: > >00493511111111 calls 00493512222222: >OpenWrt*CLI> sip show channels >Peer

Hi list! I noticed that when the phone of my wife calls the gsm codec will be used, but if someone calls the phone, alaw will be used: 00493511111111 calls 00493512222222: OpenWrt*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 192.168.200.11 00493512222222 5305ad0e07977dd 0x4 (ulaw) No