similar to: NAT and SIP.conf update.

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hi All We are unable to hear any voice where as in tcpdum it shows that RTP is flowing both ways ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and

I know Vonage doesn't officially have a "bring your own device" type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060329/5bc9f644/attachment.htm

Hi all I am trying to figuure out how to get iconnecthere incoming calls to work outbound works fine but incoming goes nowhere but to my iconnecthere vocemail if I do a sip show registry it shows up as regg'ed nnn=is my iconnect here number xxx is my secret Thank you Jeremy [general] qualify=no register=NNNNNNNNNNN:XXXX@iconnecthere/NNNNNNNNN context=default bind = 0.0.0.0 port=5060

Sorry but what does the ACL mean and its relation to the bindaddr? Regards Bilal > > 30 jan 2009 kl. 16.59 skrev Mike: > > > hI, > > > > Trying to understand how to setup two PRIs in > sip.conf. Using > > Asterisk 1.4.23. > > > > I have a provider giving me two PRI (different rate > centers) through > > SIP. Both PRI comes in from

I have the following scenario: PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) -> Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

My Asterisk box also does NAT for internal network, and establishes site-to-site VPN tunnel(s). As a result I have several internal interfaces with private addresses on them, and only one public interface. By trial-and-error I've found out that FWD (SIP) won't work unless I disable my VPN tunnels - it would send the internal IP address to FWD's SIP server instead of public one. I

Anyone suffering an outage with them right now? I am getting the following from my box when I try to dial using them.... == No one is available to answer at this time W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/e1096325/attachment.htm

Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be

I recently upgradeded dovecot on one of our servers from version 1.0.10 to version 1.1.3. Ever since, we've been seeing occasional errors similar to this sequence (with the username and IP addresses elided): Sep 30 00:09:56 alcor dovecot: pop3-login: Login: [4954], XXXX, NNN.NNN.NN.NNN Sep 30 00:09:56 alcor dovecot: wrapper[5006]: pop3, XXXX, NNN.NNN.NN.NNN Sep 30 00:09:56 alcor

I have a remote installation that connects via IAX from my office pbx. When I call an extension on the remote pbx, after the dial period, the call is terminated. Nothing I do in configuration of that extension seems to matter: -- Executing NoOp("IAX2/netconcepts@nnn.nnn.nnn.nnn:4569-5", ""Dial 710"") in new stack -- Executing

Dear list i am using Asterisk 1.2.5 with A@H . here is my problem. if i dial a number (consider 79) i have to wait around 20 seconds before my Asteisk box response. now i want to decrease this waiting time . any idea how to do that ? thanks Salaque

Is anyone using a cisco router as an ISDN gateway with Asterisk? As you might have seen from a couple of my threads, I have been looking at Fritz! and Cologne cards, both of which require development against a specific version of asterisk/zaptel (e.g. chan_capi), which is intrusdive and causes a lag in deployment. I was thinking a better approach might be to use a seperate gateway, such as a Cisco

I see you are using cups. I had the same problem It is a cups problem. First install a RAW printer in CUPS. Second allow cups to receive jobs from a remote host. By default it doesnot. -- Groetjes/Regards Kees van Hoof

I have two servers with identical hardware ... TYAN i3210w system boards with dual intel gigabit interfaces, and a PCI intel gigabit nic. I'm running Centos 5.4, x86_64, 2.6.18-164.6.1.el5 Every other time I reboot, the nics initialize in a different order. anaconda had setup /etc/modprobe.conf with alias lines for the cards: alias eth0 e1000 alias eth1 e1000e alias eth2 e1000e However,

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

_ platform i686-pc-linux-gnu arch i686 os linux-gnu system i686, linux-gnu status major 1 minor 6.2 year 2003 month 01 day 10 language R > Until this version, I've not had to explicitly start the x11 device. Now,

I have a working system with two fxo and two fxs channels. I recenlty got an IAX2 account I would like to use also. While I have gotten the IAX2 channel to "register", it remains non functional, as the incoming calls, go nowhere and the outgoing calls attempt to go out over the ZAP channel. I can see this, via the CLI, with debugs on. I strongly suspect this is a dial plan/config

I've attempted to change my smb password on a remote NT PDS, but it always fails with resolve_name: Attempting lmhosts lookup for name SERVER<0x20> getlmhostsent: lmhost entry: 127.0.0.1 localhost resolve_name: Attempting host lookup for name SERVER<0x20> Connecting to nnn.nnn.nnn.nnn at port 139 error connecting to nnn.nnn.nnn.nnn:139 (Connection refused) unable to connect to

Hello List. I am using rsync to pull html files from a shared drive to 2 web server boxes to keep the files synchronized. However, I have a few files I do not want rsync to copy over because they contain information specific (IP address) to the box hosting them. Rsync seems to intermittently ignore the exclude statement and copies them from time to time. Is there a way to absolutely prevent