similar to: signaling problems

Hi, I have been experimenting with NAT and Asterisk a bit now. Though I have made progress along the way, I have come across the following problem. I'll really appreciate if anyone can provide me any help or pointers. Thanks! Successful Scenario: ------------------- All sorts of NAT calls are successful with full two-way media when both end points are locally subscribed users. Problem

Hi ml, I have a problem related to call parking. When on my X-Lite try to parking a call dialing #700 I don't obtain anything. I can only ear dtmf tones during conversation but not other happens. I also read in some post that only pressing # should place call in hold state but this doesn't happen on my system. Can someone help me? Thanks. Marco

If I do a supervised forward on a call (Polycom 501, Asterisk 1.2.1), all goes well until the second time I hit forward (to join the caller with the extension); then, the caller's MoH goes away (making them think they've been hung up on), and the server spits out: asterisk-cw*CLI> <-- SIP read from 10.20.2.16:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP

I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH fine. After I stop MOH on Windows Messenger, if the hungup signal could not send to *, the sip channel(e.g.,SIP/52001-08ca) for this MOH remains. Then the user trys again MOH, a new sip channel starts. And again the hugup signal can not send to *,......... When I 'stop now' from CLI> , * cleanups the remaining sip

Szenario: UA(Grandstream) => PROXY(SER) => GATEWAY(*) => PSTN After sending the SIP ACK From Gateway (*) ACK sip:123456@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK86c0bd474ea746b5 From: "Me" <sip:123456@mydomain.de>;tag=0f63d269bc25545d To: <sip:100@mydomain.de>;tag=as05df60b5 Contact: <sip:100@192.168.0.1> Call-ID:

Hello, this is my first post to the list. I have several mailservers running courier-imap, but am planning to convert one (and maybe more) to dovecot 1.1.4. In the process I am also looking to finally implement shared maildirs. As it is well known some clients (notable Apple's Mail) have problems in showing shared folders if the correct private namespace (INBOX) is set in the client,

Hi, I have the following symptoms: Call X-lite / Nokia E51 X-lite press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH Call X-lite / SCCP phone MOH works as supposed Call SCCP phone / Nokia E51 SCCP press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH In addition, the BLF on the SCCP phones does NOT show the hinted SIP extension on hold. With 1.4

Hi, Are there any special statistical tests, or functions in R I could use to measure "cone-like" distributions? I have several data-sets, which I've been plotting parts of as 2D plots, where I get a "cone-like" distribution of the data points. That is, the data appears to be bounded by two non-parallel lines, starting at the origin, giving rise to a

Type 'mpg123' at the Linux CL. (no quotes) If the version is anything other than 59r, you problem is solved. Go to the Wiki and search for Music On Hold. Do the install of version 59r ONLY as described in the docs. Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Neil A. Hillard

For canreinvite=yes to work, I think I need to remove the t argument in the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways stay in the middle. I don't want that, so I removed the 't' argument. That works. Now, when two UA are calling, Asterisk gets out of the RTP stream. However, when removing the 't' argument, the Music On Hold doesn't work anymore

Folks, I have made an inbound call. Now I want to make an out bound. I have a sip softphone hitting the pbx. On my motorola firewall I have allowed all traffic in and out to my address. There is not a firewall on the PBX yet. here is the log off the softphone..and the PBX logs look like this Registration from 'Tommy Denton <sip:201@209.101.93.30>' failed for '

Thanks for the reply Guss, So in my case the Amazon VPS server is behind a FULL CONE NAT(static NAT), so is that enough to hole punch other nodes. what all ports will should be opened up on the NAT device? Most of the other nodes are behind cg-nat, so I wont have any control over its nat. Thank you, Regards, Bobby Thomas. On Oct 19, 2015 1:45 AM, "Guus Sliepen" <guus at

Hi, I recently encountered a weired situation where my budgetone stopped working. My network is like this: Asterisk on Public IP ----------- ADSL NAT Router ----- GS01, GS02, GS03 on Internal IP We have an Asterisk server running with a public IP address, which serves as the master PBX. On a remote site, we have 3 Budgetones all having internal IP addresses assigned by the ADSL NAT router. The

Hi, I have already set canreinvite=no in the sip.conf and also used the NAT=yes. But the funny thing that was in one case the user call and it wasn't working (one way audio as described) using an online dialer and then tried again using X-lite it was working. Then hanged up and tried X-lite again, it was not working. The second call was only a few seconds apart. Moving back to the online

Cool, I'll have a look at this too, thanks All! ___________________________________________________________________________________________ Mario Pio Russo, System Admin SWG IT Services Dublin, Phone & FAX: +353 1 815 2236, eMail: mariopiorusso at ie.ibm.com IBM Ireland Product Distribution Limited registered in Ireland with number 92815. Registered Office: IBM House, Shelbourne Road,

Thanks Sven, good Idea let's see if i am getting this right: 1) use MS ADSI editor to add few more attributes to the "users" class 2) use ldapmodify from my ubuntu server to populate those attributes would that work? ___________________________________________________________________________________________ Mario Pio Russo, System Admin SWG IT Services Dublin, Phone & FAX:

Hi, I have only recently become aware of Dovecot and gave it a try. The previous 0.99.9.1 version didn't work well for me (OpenSSL), I dropped it, but 0.99.10 has come just in time (saw it on freshmeat) and I thought I'd give it another try if it promised SSL fixes, and it's sorta working for me (i. e. it works with mutt, Mozilla, sylpheed, but not cone -- but cone is beta and has SSL

please reply to hdgbyi@public.guangzhou.gd.cn or bgpgong@hotmail.com, thank you. UNCERTAINTY PRINCIPLE IS UNTENABLE By reanalysing the experiment of Heisenberg Gamma-Ray Microscope and one of ideal experiment from which uncertainty principle is derived , it is found that actually uncertainty principle can not be obtained from these two ideal experiments . And it is found that

please reply to hdgbyi@public.guangzhou.gd.cn or bcpgong@hotmail.com, thank you. UNCERTAINTY PRINCIPLE IS UNTENABLE By reanalysing the experiment of Heisenberg Gamma-Ray Microscope and one of ideal experiment from which uncertainty principle is derived , it is found that actually uncertainty principle can not be obtained from these two ideal experiments . And it is found that

Thanks, I will start reading today. [image: Logo - ZUP] Thiago Anderson / Analista de Infraestrutura [image: ?cone de e-mail] thiago.santos at zup.com.br [image: ?cone de celular] (34)98857-4909 [image: ?cone de telefone] (34) 3210-8181ZUP IT INNOVATION Avenida Nicomedes Alves dos Santos, 1205, Sala 105 Uberl?ndia - MG Em qua, 16 de out de 2019 ?s 04:43, L.P.H. van Belle via samba < samba at