similar to: audio from soft phone actual phone from cloud

I am using asterisk 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I see no errors in the CLI - looks just like the previous call as far as I can tell. Then reboot the phone and as soon as its ready call it

I have a couple asterisk boxes, running sip between both boxes. 1.4.43 on both. both are installed from source, both have default settings, My config for one box is: [devgeis] type=friend defaultname=devgeis username=devgeis secret=yes disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 host=192.168.1.8 context=panel The other box is the same. There

Hi All, I have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just fine. However a call coming into the cloud asterisk answers fine - get the autoattendant, enter the extension and the polycom does not ring. The CLI shows that the correct SIP extension is being Dialed (SIP/524) Looks like I'm getting a 401 permission

Hi everybody I'm actually facing some strange messages in smb logs: our samba server is running fine but I can see a lot of this message in the log : Apr 27 13:33:53 lnx-ds01 smbd[18773]: [2009/04/27 13:33:53, 0] smbd/nttrans.c:call_nt_transact_ioctl(2029) Apr 27 13:33:53 lnx-ds01 smbd[18773]: call_nt_transact_ioctl(0x90060): Currently not implemented. after sometime (one hour, one

Hi I am running the "samba-tool domain classicupgrade", and after solving some problems (thread http://lists.samba.org/archive/samba/2013-January/170777.html), now I am getting this error: # samba-tool domain classicupgrade --dbdir ~/sambav3 --realm XXXXXX.YYYYYY.TEST --use-xattrs=yes ~/sambav3/smb.conf -d9 ... init_sam_from_ldap: Entry found for user: XXXXXX init_sam_from_ldap: Entry

Hi, I'm running dsync migrations using imapc and the source IMAP server is just too slow. It has taken 8 hours to migrate a mailbox with 47,000 messages. It seems most of the mailboxes are never accessed with IMAP, so it would be just as good to migrate them via POP3 which should work faster in my case. Is this actually meant to work? doveadm -D -o pop3c_user=xxxxxx at example.com -o

I have configured iax.conf and extensions.conf as instructed on FWD website (http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76) and I can successfully receive calls and make test calls to 612, 613, etc. The problem is that I can not make a call to another FWD user. Here is what asterisk says: -- Executing [393xxxxxx@default:1] Set("Zap/1-1",

Hi, I've got a relatively simple tinc setup. I've got two "servers" that are on the public internet that act as routers for three "clients" that are behind NATs. Those servers are called aaaaa and bbbbb the clients are xxxxx, yyyyy and zzzzz Unfortunatly the servers have problems accepting a connection from the clients syslog on aaaaa: Sep 29 18:28:58 schuerrer

I've been running a couple Centos5 and RHEL4/5 servers with samba for a while now and everything has been working great with our Windows 2003 AD. All of a sudden though I'm experience something really weird on one of the RHEL5 boxes. Whenever I try to connect as a Domain Admin from one particular Vista client, I get access denied and repeated prompts for a username/password - this has

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

Hello users, I am trying to integrate asterisk and gtalk. my configuration is as follows OS:centos asterisk-1.6.0 asterisk-addons-1.6.0 dahdi-linux-2.2 dahdi-tools-2.2 libpri-1.4 share iksemel-1.2 #/etc/asterisk/jabber.conf [general] debug=yes autoprune=no autoregister=no [google] type=client serverhost=talk.google.com username=XXXX at gmail.com secret=xxxxx port=5222 usetls=yes usesasl=yes

Hi all- Equipment: Xlite softphone Polycom 301 with SIP 2.1.1 and BootROM 3.2.3 Polycom 501 with SIP 2.1.1 and BootROM 3.2.3 Asterisk 1.4.2 SIP Trunk to FWD I wanted to post this problem as I haven't found it described in any of the past presence threads on here. I use an identical configs for a Polycom 501 and 301. (I actually unplug one when the other is in use). The

Hello, I noticed a strange thing while testing my Asterisk-Kamailio Realtime setup. In an INVITE the From and To headers contain the same number when calling through a Realtime integration setup. This happens when the INVITE leaves Asterisk. Can you guys tell me what might be causing this? I have 660 at testers.com as a websocket client and 700 at testers.com (caller) using a Zoiper client (db

Ok, I must be really stupid here - I'm playing with ael and svn trunk. given the following in ael: context isdn10 { 444601 => { Answer(); NoOp(${CALLERIDNUM}); Hangup(); }; }; isdn10 is the incoming isdn context. why do I get this on the console: -- Accepting call from '01702xxxxxx' to 'yyyyyy' on

I'm working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages. I'm currently running Asterisk 13.0.0. Anyone have any suggestions as to what I am doing wrong? The SIP provider says the latest version of Asterisk they have

Hi, I am new here on this list, and have a problem of which I hope that somebody here can help me with it. I have a Voipbuster account, with which I would like to make phone calls via my Asterisk PBX. If I let X-Lite register directly at voipbuster.com, everything is OK, but if I let Asterisk register there, it says "registration for XXXXXX@sip.voipbuster.com timed out, trying again",

I run a sshd host solely to allow employees to tunnel secure connections to our internal hosts. Some of which do not support encrypted protocols. These connections are chroot'ed via the following in /etc/ssh/sshd_config Match Group !wheel,!xxxxxx,yyyyy AllowTcpForwarding yes ChrootDirectory /home/yyyyy X11Forwarding yes Where external users belong to group yyyyy (primary). We

Hello all, Here is my problem, I try to place a call to FWD (free world dialup) trough my asterisk PBX. my config is as follow: extensions.conf ---------------- [internal] exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD) exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD

I have some stuck channels that I think I'm going to have to bounce Asterisk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels:

Hello, Anyone else experiencing problems with the latest spandsp (pre3) and last libtiff beta? I'm getting 8 bytes long file, with the TIFF header only during such connection: -- Accepting call from 'XXXXXXX' to 'YYYYYY' on channel 0/2, span 1 -- Executing SetVar("Zap/2-1", "FAXFILE=/tmp/foch.tif") in new stack -- Executing