similar to: Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11

Am 07.01.2016 um 10:55 schrieb Frank: > On Wed, 2016-01-06 at 17:03 +0100, Juergen Sauer wrote: Thx, 4answer. :) >> with in my sip.conf, I have got for this hardphone: >> [...] >> [hard1] >> username=hard1 >> secret=correct-and-three-times-checked-4-digit-pin > > In most cases, there is no need to set the "username=" option. The

Hello, please immediately Sorry, I do not know the English translation made by google. Encountered the following problem in the Samba. There is a server that is spinning Samba, there are three general spheres, and each user has a curve ball that is visible only to him, they are connected through this configuration include: [global] workgroup = WORKGROUP server string = FS encrypt passwords = yes

<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target channel found for 7792. If I'm dialing *87792 instead

<--- SIP read from 192.168.32.245:5060 ---> SIP/2.0 481 CallLeg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.32.16:5060;branch=z9hG4bK7508a694;rport From: "asterisk"<sip:asterisk at 192.168.32.16>;tag=as2ff08179 To: <sip:5386 at 192.168.32.245:5060;user=phone>;tag=c0a80101-2ce1bc03 Call-ID: 2fa28b4-c0a80101-d-9acc at 192.168.32.245 CSeq: 143 NOTIFY

MWI-related SUBSCRIBE message to send NOTIFY messages changing phones MWI status. This is fine for me but I'm wondering what if I were using SIP hardphones refusing any such NOTIFY without prior SUBSCRIBE (does such phones exist ?) ? 1. In this case, which URI shall use a hardphone to build its SUBSCRIBE message ? Here is a hand written example. Which value should I substitute to foo (in this

Hello Luca, We are still playing with visualization of your data, but I didn't want you to wait any longer for some results.  I think I blame both DT and the Pi :) First, a look at the phone side of your Banana Pi.  The first thing we noticed is there were a LOT more packets in one direction (north towards DT) than the other (towards the phone): jeff at

Hi, I installed Wine 0.9.30 using Automatix 2 onto Xubuntu Edgy primarily so I could run IE 5-6 for testing purposes. I then installed IEs4Linux. It all works fine, although I still feel a sense of horror to see IE on my linux desktop, but the title bars are all messed up for both winecfg and IE. The link below shows a screenshot of the effect: http://people.aapt.net.au/~adjlstrong/wine.png

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 :

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

Hello, When sending this AMI request ... 192.168.64.5 -> Action: Originate 192.168.64.5 -> Channel: SIP/9122 192.168.64.5 -> Async: True 192.168.64.5 -> Callerid: 9122 Guest2 <9122> 192.168.64.5 -> Exten: 9123 192.168.64.5 -> Context: local 192.168.64.5 -> Priority: 1 ... I've got this INVITE from Asterisk INVITE sip:9121 at 192.168.100.198:5060;user=phone SIP/2.0

Hello, I'm using Thomson/Technicolor ST2030S hardphones with Asterisk 1.6. Changing from 1.6.1.18 to 1.6.2.6, I can see a change in Pickup's behaviour and I'm a bit confused about it. With 1.6.2.6, when extension 7791 is calling extension 7792, I can see INVITE messages coming in and out Asterisk. I can also see a NOTIFY message advertising this call to subscriber 7793, for instance.

Hi, I have an Alteon in test (a sip/rtp load balancer). This Alteon sends to the asterisk box a "SIP OPTIONS" to know if asterisk is alive. However, asterisk sends me a 404 message and not a response like, for example, a Thomson (200 + SDP) I wrote a very little script (you can find it at the end of the email) to send an Options message to asterisk/phones to try. It works

Hi. I'm hoping that someone on this list can help me with my problem. I've been looking on my own for a question for the past few hours at least, so hopefully this isn't just a FAQ. Anyhow, my problem is simple (to describe). The NIC's boot agent gets an IP address (verified with dhcpd), gets the pxelinux boot image and correct configuration file (verified with tftpd), and then

Hello, I've got strange problems trying to run asterisk with MGCP ip phones (Thomson ST2030). Situation: "user A" <----- pstn -------> ASTERISK <----- mgcp ------> "user B" "User A", connected behind a PSTN, tries to call "User B". After dialing "User B"'s number, call comes to ASTERISK, ASTERISK contacts

hi there, I saw a page on voip-info about the thomson ST2030 phone. There is not so much info on there, that's why I would like to raise a question here. Has anyone got hands-on experience with this phone? (with or without extension module) I am interested if it can be used (as SIP phone) in a good way with asterisk. (also, do all the functions behave like they should; like Supervision

Hello, After reading all the docs and going through the menus, I still can't find the voicemail access button or menu sequence on the ST2030 (http://www.voip-info.org/wiki/view/Thomson+ST2030) Also I can't get phone provisionning through tftp to work. Configuration files are loaded but the phone seems to ignore them. Any idea?

Hello. Once again, I came up with a problem for which I can't seem to find a solution. I'm not able to make BLF work with Thomson ST2030 phones and Asterisk (1.2.13). I've set up hints in dialplan, as well as Subscibe keys on the phone. The LED status gets updated according to the associated line status. However, when a phone is ringing, If I try to pickup the call by pressing the

Hello, I'm trying to upgrade a Thomson ST2030 phone froms its default 1.42 firmware to the latest version (1.56) through tftp. The phone loads the .inf file, then the correct firmware file (as stated in the ST2030S.inf), then it reboots and loops doing these same things again and again. The firmware version on the phone stays at 1.42. Is there a special intermediate firmware version to

Hy all of you, I've successfully installed a freepbx solution with 10 extensions : - 5 on Linksys SPA922 - 1 on Linksys SPA942 - 1 on Thomson ST022 Everything seems to work fine with all the hardphones excepts last week. The thomson has a strange behaviour. It can reach french mobile cell phones but when it reaches "fix" phones, the correspondant can't hear the caller. What

Taken from http://www.twice.com/html/pagebeta.cfm?InputKey=2853 ------------------------------------------------------------------------- Thomson, Fraunhofer Developing Enhanced MP3 Codec Called mp3PRO Jan. 7, 2001 By Joseph Palenchar Las Vegas -- Thomson and the Fraunhofer Institute, the co-developers of MP3, are working with a third company to develop an improved MP3 codec intended to deliver