similar to: Choosing codecs

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! 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

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > 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 :

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

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'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

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

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: > 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

> 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]:

Ishfaq Malik <ish at pack-net.co.uk> schrieb: Hi Ishfaq > Look into Busy Lamp Field/Presence > > Here's a starting point: > > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Thanks a lot, but it does not seems to work... Here my configuration: sip.conf: [general] allowsubscribe=yes subscribecontext = default

Am 13.06.2020 09:30, schrieb Luca Bertoncello: Hi again (again) I noticed right now another strange detail... I made a call using my mobile phone (connected to the Asterisk). The quality was top... Maybe is the problem in a codec used from our phones at homes? Could someone suggest me how to check the codec used by my mobile phone and the codec used by the phones at home? Thanks Luca

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Monday 06 Jul 2015, Luca Bertoncello wrote: >> Well, but for voice quality, which codec is better? >> alaw or gsm? > > A-law is better for voice quality (sorry, thought my original > explanation was > obvious). But note that if the destination is a mobile phone, GSM will be > used anyway, at

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: Hi, > GSM is the native codec used for calls to mobile phones; it uses lossy > compression to achieve a low bit rate. > > A-law is the native codec used by physical exchanges on the land line network > (PSTN and ISDN). It is non-lossy. It works by arranging the "steps" closer > together near the zero

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > Yes. You should definitely be using A-law for calls to the Outside World. Well, I wanted to change these settings, but I'm not sure, where I have to do that... I think in the users.conf, but I think, the "allow" keywords is for the network... How can I change this setting? Thanks Luca Bertoncello (lucabert

> 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

Hi again! I always try to get my mobile phone work with my Asterisk. I tried to install Asterisk on my PC (with public IP), but it has problems, too... I think, my UMTS-Provider doesn't want to connect to dynamic IP or my DSL-Provider does not want it, too, since I have no problem to connect and get a very good audio quality if I connect to other SIP-Provider or to an Asterisk (SAME

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

Dear list, I am trying to re-analyse something. I do have two time series, one of which (ts.mar) might help explaining the other (ts.anr). In the original analysis, no-one seems to have cared about the data being time-series and they just did OLS. This yielded a strong positive correlation. I want to know if this correlation is still as strong when the autocorrelations are taken into account.