similar to: SIP Nat Issue

Hi everyone, Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing. Tomica ---- This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr

Hi everyone, I want to have a group and dial multiple phones/lines simultaneously. If I use this Dial command: exten => 222,2,Dial(${TRUNKBP}/246&SIP/258&${TRUNKBP}/243,20,tTr) ... all phones ring just once, after that only the first one continues ringing and only that one can answer. Can anyone tell me why? thanks! Tomica -------------- next part -------------- An HTML

Can someone tell me what is wrong here: Feb 2 19:45:44 ERROR[1074441696]: cdr_addon_mysql.c:381 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. The database is created, cdr table also, the username and password is right. I have tried configuring cdr_mysql.conf to connect via localhost mysql.sock or via tcp port, but in both cases I got this error. Thanks!

has anyone seen this? -- Starting Skinny session from 192.168.13.102 -- Starting Skinny session from 192.168.13.102 triton*CLI> Oct 15 13:44:05 WARNING[213019]: File chan_skinny.c, Line 2243 (get_input): Skinny Client sent less data than expected. Oct 15 13:44:05 NOTICE[213019]: File chan_skinny.c, Line 2301 (skinny_session): Skinny Session returned: Success Oct 15 13:44:05

I don't know if this is a problem with my cisco sip IP Phones or asterisk but I thought I would post here in case someone else has experienced this issue. When I make a call from my SIP cisco IP Phone to some remote IVRs I never get the rest of my soft keys, only the "End Call" soft key, and also DTMF doesn't work... its like the phone is acting like the remote end hasn't

Hi, I have a Cisco 7920 that I'm trying to get working with my * box. When the phone boots it requests XMLDefault.cnf.xml and SEP<MACADDRESSHERE>.cnf. I assume I set the line number, etc in the latter of the two. However I cannot find any reference to how this file is structured. Anyone know? I assume this is why I'm getting the errors below: Oct 17 19:47:24

Hi everyone! Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a

Hi everyone, Maybe someone could help me. I have Asterisk in production with TE410P connected to PSTN. When I call from internal phones, either voip or connected via other PRI trunk, to PSTN and if the called phone is busy I don't hear anything!?! I should hear tone indicating that called number is busy. I have played with busydetect and callprogress in zapata.conf, but I didn't find

Hi, I wonder if you guys can recomend a good SIP phone. A phone thats works great with * has a lot of features, and is cheap. Actually all kind pf VoIP hardware is of interesst. Is there a really good site for VoIP harware ? /Mike

Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone

I need a good reliable source for channel banks. Ebay is nice but too many variations of quality. I need one that can do 8 FXO and 16 FxS does anyone have a particular model, other than adtran that they can recommend??? And if possible include a source to buy from... Thanks in advance, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL:

Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url :

hi, I saw that Mike Dransfield tried to port 'state' plugin from beryl. What does it do? It should be able to place windows, based on name, class etc., to specific viewports. I recently converted to compiz window manager from WindowMaker (used it for 8 years), and I miss automatic 'pinning' of specific windows to particular workspace (or viewport in this case). Mike's

>I went back to the example system direct from CVS with small >additions to sip.conf and extnsion.conf needed to make one >xten X-Lite phone work. I can dail in and hear the anouncements, >call the demo server at Digium. The audio quality I hear >comming from Asterisk back to X-Lite is good (9 on a 10 scale) >but the sound volume comming from the X-Lite extension is very low

Hi U can visit the http://iaxclient.sf.net for some opensource underdevelopment softphones. Take Care Obaid Amin Syed >From: Chris Albertson <chrisalbertson90278@yahoo.com> >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] Good W2K softphone >Date: Fri, 3 Oct 2003 23:00:13 -0700 (PDT) > > >I haven't

Hi, My installation at home use two NAT translations before it reaches the linux box where Asterisk is running on. I use DSL with a Wireless router which fwd all packets to an Windows 2003 box an this windows box it NATing the UDP and RTC packets to my linux box. If I try to connect to it from outside I get this error : Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum

I have just checked the Openbsd box on the if interface. fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xffffff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0:

On the cheap side, the ITX or even MicroATX machines work great. These are commodity items, so they tend to be far less expensive than custom solutions. Various manufacturers, but we've had very good success with any of the AOpen MicroATX boards and their slimline MicroATX case: Aluminum: http://usa.aopen.com/products/housing/A340-series.htm Steel:

Hi all, Can any one please help me in intergrating PHP/Mysql with my running asterisk server to configure IAX or SIP users? I will highly appreciate any help in this regard. Regards Nawaz. --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, >

I see with the introduction of CentOS 7.5 there's a new qemu-kvm-ma package on ppc64le (which is actually newer than qemu-kvm-ev currently). Does anyone know what the difference is between these two packages? We currently use qemu-kvm-ev and we've run into this bug [1] which got me wondering if we should be switching to that package on ppc64le. Thanks! [1]