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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.

<--- 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

Hi! I wish you all e Happy New Year first! Allthough, I'm relative new to Asterisk, I got our server up and Running, Softphones, ISDN, and a brand new Snom 821 are working flawlessly. :) Platform is Debian 8/Asterisk Packages (11) from Debian Repo. But I am running into problems setting up 2 older Hardphones, Thomson 2030S. :( with in my sip.conf, I have got for this hardphone: [...]

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 would like to test the following COLP use case : Alice and Bob are both using a SIP phone registered on a Asterisk 10 server. Alice dials Bob's extension. While Bob's phone is ringing, Asterisk updates Alice phone screen with Bob's name, so that at a glance, Alice can check she dialed the correct number. Before diving into Asterisk documentation, I would be happy to be

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, I'm banging my head over this. Usually, I'm using a SIP hardphone feature called "Call Pickup Starcode" to enhance BLF with Directed Call Pickup : basically, SIP hardphone (here a Thomson ST2030S) is configured to send an INVITE message whenever a BLF is pressed while blinking. The INVITE is build with the extension number (attached to the BLF that was blinking and pressed)

I am trying to use asterisk for an shared line gateway. When moving from one phone by placing the call on hold then having a second phone pickup that held call by sending asterisk a replaces header (http://www.ietf.org/rfc/rfc3891.txt) Asterisk does not seem to send a "bye" message to the original UA leaving the first phone stuck in a holding state. Am I missing something here? Here

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

for the same price, siemens dect phones with sms feature display caller name !!!! --------- Message d'origine -------- De: Wilson Pickett <spamsucks2005@gmail.com> A: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Objet: Re: [Asterisk-Users] Major problems with TDM400 and specific Date: 25/03/05 18:00 > > &gt; Sorry for my bad

Hi, I've a huge problem with the following: Setup: Asterisk 1.4.11 I've got two Thomson ST2030s in an queue. After a while Asterisk logs the following if somebody calls the queues number: - Got SIP response 486 "Busy Here" back from 172.10.3.31 -- SIP/office1-0823d190 is busy -- Nobody picked up in 0 ms The phones are NOT busy (show channels show nothing). Also

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

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

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

Here's another question related to how to use masks -- In my PDC area I specify: logon path = \\netapp\profiles\%u This puts server-based (roaming) profiles on my Network Appliance (which itself is an SMB/PDC client). A previous admin here left this commented section: #[profiles] # path = /var/lib/samba/profiles # path = /netapp/profiles ??? # read only = no # create

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

major undertaking. But since you are using an AGI to control the Queue command instead of using it from the dialplan, you have more control over this problem than you realize. For simplicity of illustration, let's say your AGI simply wants to take a call and send it to the next agent in the queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the Polycom transfer from

used Kamalio to "supplement" the features that Asterisk either doesn't provide or doesn't provide in as nice a form as the OP desired - can't really speak beyond this as I am not one of them. ------=_NextPart_000_010C_01CB6EAA.3AC2C610 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <html

Jan 19 10:00:29 VERBOSE [7177] logger.c: -- Executing [1000 at ext-meetme:7] Read("DAHDI/2-1", "PIN|enter-conf-pin-number||||") in new stack Jan 19 10:00:29 VERBOSE [7177] logger.c: -- <DAHDI/2-1> Playing 'enter-conf-pin-number' (language 'en') Jan 19 10:00:43 VERBOSE [7177] logger.c: -- User entered