similar to: Problem with 401 Unauthorized

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me

On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com> wrote: > So I've got a bit further with my project to get BLF working between > asterisk and linphone. > > Initially asterisk was rejecting linphone's SUBSCRIBE messages because > they didn't have an Accept: header. I've fixed that and now the initial > SUBSCRIBE messages work and I see all my

> On 23/03/2020 18:51, Joshua C. Colp wrote: >> On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com >> <mailto:john at calva.com>> wrote: >> >> >> Why is asterisk giving an error 500? I can find no reason, there >> is nothing in any log. >> >> >> The sequence number is from the past. The first SUBSCRIBE is

Hi John, 1. Could you get any further, in your quest for working BLF with linphone ? 2. Have you tried with a different Linphone version (4.12 is pending on Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ? Best regards Le mer. 25 mars 2020 à 15:06, John Hughes <john at calva.com> a écrit : > > On 23/03/2020 18:51, Joshua C. Colp wrote: > > On Mon, Mar

So I've got a bit further with my  project to get BLF working between asterisk and linphone. Initially asterisk was rejecting linphone's SUBSCRIBE messages because they didn't have an Accept: header. I've fixed that and now the initial SUBSCRIBE messages work and I see all my online contacts in green. But after a few minutes linphone attempts to renew the subscriptions and

Hello, I'm having difficulty with registering a SIP account in a Snom 320 IP-phone. This is what sip debug tells me : [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] <--- SIP read from UDP:public_ip:58697 ---> REGISTER sip:sip.domain.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.114.200:2048;branch=z9hG4bK-vj1xvbdnp4dw;rport From: <sip:test3 at

Hi, I'm trying to register SIP phone with an asterisk serve, failing miserably. The server is sending "401 Unauthorized" responses to the registration attempts, but every time the phone is re-REGISTERing without authorization. I'd think this was a problem with the IP phone, except... the very same phone registers correctly (authenticated) with another asterisk box, same brand,

Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context |

Hello everyone, A few days ago I had a problem with a couple of extensions. I have about 12 Aastra 6731i phones, 6 are at our main office and 6 more on remote branches. We use VPN to communicate to our branches so there's no NAT involved any where. The problem I had was that I couldn't call from two extensions located at two branch offices. But I could call to them just fine. On any call

??? I am having a really bad day trying to get incoming calls to work on Asterisk 13 with PJSIP.? We just migrated from Asterisk 1.8 where everything was working but there seems that something got lost in translation.? No matter what I try I always get a 401 Unauthorized message when receiving a call from the PSTN provider.? I can make calls and the registration is working.? I have tried to

I am unable to get any softphone to register to my asterisk server when I am connected via VPN. I have tried Ekiga, LinPhone, and Twinkle... on multiple machines. It works fine when locally connected (same subnet). The VPN is not NAT'ing anything... and all other connections work fine across it (i.e. http, ssh, scp, ftp, etc). In fact, the asterisk logs show the connections, so its getting

On Mon, Mar 23, 2020 at 7:15 AM John Hughes <john at calva.com> wrote: > Hi, in these dark days of COVID-19 lockdown I'm using linphone to > connect to my office asterisk system for working from home. > > It's going pretty well but the presence/BLF functions don't appear to work. > > In the linphone logs and asterisk debug I find that asterisk is > rejecting

I am getting a SIP 401 unauthorized message. My public IP or PIP is being pre-routed with iptables to goto an internal IP or IIP All the polycom phones in the office point to the IIP. they work fine. I have 2 external phones that are registering to the PIP. I see the register attempt as I am getting the 401 unauthorized message. For the 2 external phones both have nat=1 enabled. remote phone

Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context | secret |

Hi, in these dark days of COVID-19 lockdown I'm using linphone to connect to my office asterisk system for working from home. It's going pretty well but the presence/BLF functions don't appear to work. In the linphone logs and asterisk debug I find that asterisk is rejecting linphone's PUBLISH message: <--- SIP read from UDP:10.27.128.3:5060 ---> PUBLISH sip:john at

I'm experiencing one-way voice paths, followed by a hangup on one softphoine and not the other. Also, caller ID has odd outputs -- and I wonder if the problems are related. My configuration has Asterisk and a Linphone softphone running on the same box. I have a PC, and on that PC I use X-Lite or SJPhone to connect to the Linphone instance. When I call from the PC to Linphone: * I call

It seems a new Linphone 4.2 is to be published next week ! Hopefully, ... Le ven. 5 juin 2020 à 13:34, John Hughes <john at calva.com> a écrit : > On 26/05/2020 15:33, Olivier wrote: > > Hi John, > > 1. Could you get any further, in your quest for working BLF with linphone ? > > The patches to get linphone-3.12 BLF working with Asterisk are here: > >

I'm trying to register with net2phone. I've already changed chan_sip.c, User-Agent: string to say "User-Agent: Cisco ATA 186 v2.16 ata18x (030401a)". But still I'm getting the error msg. Here is the debug msg: IP Address is xxx.xxx.xxx.xxx 11 headers, 0 lines Reliably Transmitting: REGISTER sip:66.33.146.12 SIP/2.0 Via: SIP/2.0/UDP

On 26/05/2020 15:33, Olivier wrote: > Hi John, > > 1. Could you get any further, in your quest for working BLF with > linphone ? The patches to get linphone-3.12 BLF working with Asterisk are here: http://perso.calvaedi.com/~john/linphone-3/ They're pretty damnned trivial: 1. add the "Accept" header to the SUBSCRIBE message so asterisk doesn't reject it. 2.

Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten