similar to: asterisk-users Digest, Vol 58, Issue 9

<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

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.

Hello! i'm trying to set up transfer without using the respective asterisk-function but with the built-in phone functions. my goal is to have the first callleg billed to the caller and the second callleg to the callee, who is responsible for the forward(and i can't bill a unknown caller anyways) so far it's working without problems, but my cdr's are messed. with the help of the

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

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

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)

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

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

Please help find the cause of strange behavior res_pjsip. Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up! Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d No diagnostic messages why asterisk suddenly decided to hangup i don't found :( There are suggestions or strong belief

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. 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

Hi guys,

Hi everybody, quite frequently I build customized RPMs with asterisk-1.4.20.1 including some special patches for it, to install the on CentOS 5. Now I was looking to upgrade to asterisk-1.4.24.1, but the RPM-build is not working anymore with my build environement. In version 1.4.22 the "Makefile" was modified and all the RPM-stuff was removed, same for the