similar to: Busy problem

<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

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

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

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)

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. 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 all, Im using Asterisk 1.4.11 and I want to proceed some time and date operations in my dial plan. (for a time shifted callback). Should look like: CURRENT TIME + x minutes. Of course it should increase the hours for example in this case: 10.59 + 5 minutes = 11.04 I guess I've to use the math function in 1.4 but how can I manage easily the time operations? Kind Regards, Erik

<--- 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

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, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on .... not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) Thanks! Kind Regards, Erik

Hi, Is there a possibility to control sound levels (higher / lower) in Asterisk (so the codecs). Somebody asked me to evaluate that but I didn`t found any documentation about. I have the opinion that for these (audio) things the end user client is the only part where I can tune around. Problem is for example a (Austria) ISDN --> Asterisk --> SIP / IP ---> (Romania) Asterisk

Hi, Im sure somebody out there had the same "problem before. IF a call comes in with suppressed caller id (Call Centers, etc.) 'asterisk' is shown as CallerID. Can I change somewhere this behaviour to display like ' Unknown' ? Thanks! Kind Regards, Erik

I have several branch offices all running Asterisk PBX's that register to each other via SIP so that calls can be transferred from office to office. Everything is working great on the office to office transfers, but I'd like to somehow make the CallerID more useful. Currently if an extension at Office1 dials an extension at Office2 the CID on the phone at Office2 says

Hey all, I hope I'm missing something simple. This is my second PDC install and I'm having some problems getting my win2k machine to join the domain. First I made the machine account: useradd -g 100 -d /dev/null -c "bob's computer" -s /bin/false office1$ Then I lock the password: passwd -l office1$ Next I make the smbpasswd -am office1 name. Finally, I added a

I have several branch offices, each with their own Asterisk server (version 1.4.22.1) handling their PBX functions. All of these offices need to talk to each other. In sip.conf I created a peer entry for each office with a username of branch-user and a friend entry for every branch-user with the username being just the branch, for example: [Office2] username=Office1-user host=10.10.80.253

Hi, I need a feature from asterisk with dahdi channels, if there is an incoming call, it should ring on several dahdi channels. My channels look like: OFFICE1=DAHDI/13,,rtT OFFICE2=DAHDI/14,,rtT If I add this line: exten => 12345678,1,Dial(${OFFICE1}&{OFFICE2}) only OFFICE1 rings. If I change it to exten => 12345678,1,Dial(DAHDI/13&DAHDI/14) DAHDI/13 and 14 rings together,

I'm new to asterisk and need some help with getting a SIP connection working. I am trying to establish a termination point/DID number in another country. I am currently running Asterisk CVS-HEAD. My foreign provider uses SIP and authenticates via IP address. I am not required to register my SIP connection in order to send or receive calls. Can someone help me with how to understand the

Hi Folks, First of all, I've been reading a lot and a lot about asterisk and ISDN-BRI, but is really a complicated world, not like PRI or Analog. I have 2 offices with 2 NT boxes and I want to connect them with Asterisk to route outgoing from one office two the other. At the end I want them to look like this Office2 will have only VoIP phones (not ISDN phones connected to a card in asterisk)

Hi, I was thinking about a common (I hope!) use case of Glusterfs geo-replication. Imagine 3 different facility having their own glusterfs deployment: * central-office * remote-office1 * remote-office2 Every client mount their local glusterfs deployment and write files (i.e.: user A deposit a PDF document on remote-office2), and it get replicated to the central-office glusterfs volume as soon

Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty