similar to: Dial timeout and SIP 302 Moved Temporarily

-- On Friday, November 21, 2003 12:48 PM -0800 "David Wolfskill - david@catwhisker.org" <+freebsd-security+openmacnews+0459602105.david#catwhisker.org@spamgourmet.com> wrote: David, thanks for your reply! >> i've been struggling with setting appropriate rules for an SMTP-server >> behind by NAT'd firewall. > > OK.... <snip> > >>

Hi all, I want to two sip clients connect through Asterisk in local network for testing. My sip.conf file looks like this [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=192.168.1.0/255.255.255.0 [7001] type=friend host=dynamic

Hello all! We have a PSTN line with four numbers calling into it. There is distinctive ring on these lines. They are are follows: 1. standard ring 2. short ring 3. long ring 4. short ring, long ring, short ring Based on the information I have been able to find, I have created the following entries in my zapata.conf file, to try and weed out some of the timings: dring1=95,0,0

Hi, Anyone know where i can find out more info on Reaper/Spawner. Currently, every time i add a new account on my production machine, i have to restart the whole server. After about 150 accounts, this puts a real strain on the server (it takes 3 full minutes before i can access any site on the server). I think reaper/spawner is my answer, but i am havving trouble figuring out how to use it.

Hello, I want to redirect one local port to another. I am using the following: iptables -t nat -A PREROUTING -p tcp --dport 7003 -j REDIRECT --to-ports 80 and testing it by telneting to localhost on port 7003. It works on Centos3, not on Centos4. No luck with this either: iptables -t nat -A PREROUTING -p tcp --dport 7002 -j DNAT --to 127.0.0.1:80 Am I doing something wrong? Or did something

Hi all, I use centos7 and don't want to use tmpwatch as well as crond. I have a question to use `systemd-tmpfiles-clean.service` with my custom configured file in `tmpfiles.d` to delete outdated files periodically in some log dir. I have a `tmpfiles.d` configured file in `/etc/tmpfiles.d` named `my_log.conf` in following contents. ``` #Type Path Mode UID GID Age Arg r

We''ve been noticing a lot of these errors when booting VMs since we moved to 3.0.3 - I''ve traced this to the hotplug scripts in Dom0 taking >10s to run to completion and specifically the vif-bridge script taking >=9s to plug the vif into the s/w bridge on occasion - was wondering if anyone has any insight into why it might take this long. I added some instrumentation to

Dear all, I want to make a filter for all IRC-Dalnet traffic, so I want to put all traffic for port 6660, 6661, 6662, 6663, 6664, 6665, 6666, 6667, 6668, 6669, 7000, 7001, 7002, and 8000 to a class. So, I create a TC script as below. I''m sure, it is not effective, and we can write it in simpler. I need help, how to make my script below are simpler. The simpler, the better. Thank you

Do you know where to find clear developers' guides (with some examples) for developing apps that run *on* Cisco 79xx phones (especially the 7970)? Examples that can run against Asterisk (not CallManager) with SIP firmware (not SCCP), and/or LDAP directories (or other open servers) would be best. On Sat, 2007-09-01 at 12:00 -0500, asterisk-users-request at lists.digium.com wrote: > Date:

Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see

Hi All, I am working on creating storage driver domain with xen-4.3.0 . I am referring this - http://wiki.xen.org/wiki?title=Storage_driver_domains&oldid=9371 I have created Storage domain with xl utility of xen, then followed above wiki for rest of the steps, additionally I did install blktap-dkms package also- Following xen processes are running on Domain 0:

Hi All, I am working on creating storage driver domain with xen-4.3.0 . I am referring this - http://wiki.xen.org/wiki?title=Storage_driver_domains&oldid=9371 I have created Storage domain with xl utility of xen, then followed above wiki for rest of the steps, additionally I did install blktap-dkms package also- Following xen processes are running on Domain 0:

hi all, i've been struggling with setting appropriate rules for an SMTP-server behind by NAT'd firewall. it's not that there is too little info on the web -- or here, for that matter -- there's scads of it for seemingly endless configs/req'ts -- none that seem to be exactly my own. bottom line: i'm a bit confused, and looking for some experienced advice. my goals (for

Does Asterisk 1.0.2 support 302 redirects? With 1.0.1 I get: Got SIP response 302 "Moved Temporarily" When forwarding the call to other SIP server. This is a "bug": http://lists.digium.com/pipermail/asterisk-users/2004-May/045774.html --- Jan Baggen - jbaggen@ip2.nl IP2 Internet BV / http://www.ip2.nl

Dear all, In my Asterisk 1.2.17 architecture different levels of permissions are established using different contexts that hierarchically include more permissive contexts until default context is reached. In default context there are only local extensions, only in more restricted contexts there are the PSTN access. So, if some user dials some number, Asterisk looks which context that user

When a call is transferred to another extension using a blind transfer, asterisk keeps traces of who is transferring in the BLINDTRANSFER variable. If instead the call is "forwarded" using most phone call forward feature, a 302 Moved Temporarily is sent back to asterisk -- Called SIP/104-DEVEL -- Got SIP response 302 "Moved Temporarily" back from 83.211.***.***:5063

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> Hello,<br> <br> I have an * installation that sometimes receives&nbsp; a 302 "Moved Temporarily" response to an INVITE. * sends the ACK but takes about 30 seconds to start the new INVITE

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in public network (Mobile Phone) incoming call to

I have some Cisco 7905 phones with the SIP load 1.02.00(040406A). When the phone is off-hook but no call has been placed, or when the Do Not Disturb is activated, the phone returns a 302 "Moved Temporarily" message back to asterisk as follows: ----------- -- Executing Dial("SIP/5060-0811bb00", "SIP/9871234|20|Ttr") in new stack -- Called 9871234 -- Got SIP response

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)