similar to: Call Placing timeouts

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients(using linphonec). In a proper context, I have mentioned extensions 107 as simputer@X.X.X.X (x.x.x.x=asterisk server ip) Asterisk Sever-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on

Greetings, Today, there was something on the news of the 'simputer'; a very 'simple computer' that currently is being developed in India. (http://www.simputer.org/). (palmtop computer priced some 1000 Rs -which should be roughly some 160 euro-. It's based in linux and one of the ways it tries to be as cheap as possible is by avoiding all kind of licensed technology. Now, when

Hi list, i have a problem mounting shares on a single client with diffrent usernames. if any user on the client gives a wrong password all existing cifs mounts are closed by the server. Since this problem occured in a productive environment i set up a testing env and tried to keep it as easy as can be: Servername: serverA Servicename1: home1 Servicename2: home2 Clientname: clientA Username1:

Hi, i have sporadic connection problems with my rsync server. I have _no_ firewall involved and there is _no_ max connections set. rsync on the clients is started from cron via a bash script. There's also no permission problem. rsyncd runs as daemon (is not started from inetd). It sometimes works, but every third or fourth rsync fails with this message: rsync: failed to connect to

Dear all Recently, i did some work to test small files performance for gnfsv3 transport. Following is my scenario. #####environment##### ==2 cluster nodes(nodeA/nodeB)== each is equipped with E5-2650*2, 128G memory and 10GB*2 netcard nodeA: 10.254.3.77 10.128.3.77 nodeB: 10.254.3.78 10.128.3.78 ==2 stress nodes(clientA/clientB)== each is equipped with E5-2650*2, 128G memory and 10GB*2

Hi We are provisioning some default features to our customers like "automated answer when outside labor time", "rerouting when Subscriber Absent", and so. These are macro calls embedded in key points inside the dialplan. Since not all customers need/want all features and they (the features) are customized, I named them like [macro-feature1-ClientA], [macro-feature4-ClientF],

this is the case: serverA [ CentOs 5.6 kernel 2.6.18-238.12.1.el5.centos.plus, Samba ver. 3.5.21 ] serverB [ CentOS 5.6 kernel 2.6.18-348.1.1.el5.centos.plus, Samba ver. 3.6.6-0.129.el5 ] clientA [ WindowsXP ] clientB [ WindowsXP ] The serverA shares via Samba the resource [test] [global] workgroup = DMIT netbios name = SAMBA server string = DMIT domain server interfaces = eth0

I have a asterisk server running on site listening on a public ip. Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed. My home network is simple, Cox cable connection hooked to a linksys wrt router. The firewall on the linksys router is disabled and I even setup dmz to the phones ip as a last resort. I removed the linksys router and plugged the phone

Hi, I''m looking to run several installation of a rails app. Each one has certain unique elements, like database.yml some and some css stuff, possible some templates as well. I''ve set this up like /w/sites/clienta /w/sites/clientb /w/sites/clientc Inside these directories there will be mostly symlinks to: /apps/myapp Urls for these apps will be: http://clienta.example.com

Hi, I have some problems and doubts connecting two asterisk servers. I have one asterisk (serverA), with 1 sip client registered (clientA). I have another asterisk (sever B), with another client (clientB). Now I want to call from client A to B and from B to A. Searching in google i find many configuration examples. For instance:

I've seen the following appear in some tests with Asterisk 11.11: WARNING[3938][C-00000003]: chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101 Specifically, it always happens from a Firefox 24 host but it works without this error from another host running Firefox 26 I did a diff on the SDP and couldn't see

I've attached two SIP debugs in reference to bug #116. They are from today's CVS build. 1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the call. After which, SIP(2) rings for about 30 seconds then stops. 2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before the call is answered. SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom

Hi Muhammad From: "Muhammad Talha" <talha@worldcall.net.pk> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, December 22, 2004 2:07 PM Subject: Re: [Asterisk-Users] Zaphfc/BRI Configuration help > Thanks for lan for your reply can you share your extention.conf > setting . > >

Hi. I'm stuck into an odd situation. Here's what happens: 4 Thomson ST2030S 2 Cisco 7912 3 Cisco 7940 2 AAstra 480i Asterisk 1.2.17 Diva 4BRI + chan_capi I've just upgraded yesterday from Asterisk 1.2.13 to 1.2.17. Until yesterday, everything was just fine with 1.2.13. Immediately after the upgrade, *all* the 7940 are no more able to make calls, just receive them, while 7912

i'm planning to migrate a call center to asterisk, i don't understand if i can launch a resident application on the agent's client in relation with the queue the agent's is answering. For example: I have - queue A - queue B - queue C Agent 100 (logged in A.B,C) Agent 101 (logged in C) When Agent 100 receives a call from the queue A i'd like to launch his browser and point

another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says ?unable to set up conference? and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just

Hi Guus, Am Freitag, den 25.09.2015, 17:04 +0200 schrieb Guus Sliepen: > Ok, that means by default the UDP NAT timeout on the Cisco is extremely > short. > > > I check the manual of the the Cisco NAT for any TCP/UDP > > timeout settings, but there is no way to modify anything like "keeps > > TCP/UDP connections alive". > > It wouldn't be called

I've been banging my head on this for several hours, and I have no idea what's going on. Maybe there is a very simple result, and I've been looking too hard at this this evening. This is a brand new system, and I'm wondering if there have been SIP bugs introduced in the latest CVS that are preventing from working what should be a stupendously simple test. - Cisco 7960

I've got three 7960s running v6 SIP firmware. My Asterisk setup has worked fine with grandstream devices, and basically, we're just upgrading to use nicer phones. Whilst I can make/receive calls from the 7960 to/from gossiptel). When I try to place a call, I get the following Jan 21 11:09:23 NOTICE[19688]: chan_sip.c:7271 handle_request: Failed to authenticate user "30"

I'm having a tough time getting call pickup to work on *. Here's my configuration: X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image A call comes in, and * picks up and presents a menu. Caller chooses extension, (in this case ext 103, SIP/wsmith) Wsmith is sitting in my office, hears his phone ringing, picks up my phone, gets dial tone, and presses