similar to: asterisk-users Digest, Vol 70, Issue 30

Hi Vardan I did same as you told and deleted the SIP information in Astdb and restarted asterisk. but the result was same. as you said there might be mistake in sip.conf so i am pasting both servers configuration here.. 1- nasir.server.com [abc] username=abc type=friend secret=mysecret nat=yes mailbox=12234568 incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=payasyougo

Hi, I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls). With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious. I followed the simple example at http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/. so Asterisk server 1 (192.168.250.111) sip.conf contains: [interboxsip] type=peer host=192.168.250.112

Hi, I am trying to achieve Stateless NAT with ip rule and ip route. Thanks to LARTC doc, I have done it :) But, I have a lot of client wanted access to Internet, setting up 2 rules for each of them is not desirable. For example I have 2 clients: Current setting: [root@son-ag webauth]# ip ru 0: from all lookup local 32760: from 192.168.8.113 lookup main map-to 192.168.250.113 32761:

Thanks Vardan, I will like to know if this scenario can work when peer is not having fixed ip and we use host = nasir.server.com ? also I have set insecure=invite,port what if i use insecure=no thanks again. Message: 24 Date: Tue, 11 May 2010 10:52:14 +0500 From: Vardan <hvardan71 at gmail.com> Subject: Re: [asterisk-users] Dialing a SIP Peer without using register strin To:

Yes this scenario works on my 2 systems which are at LAN. I made one system as server (192.168.0.20) and registered from other system... it is fine but now there is a different scene. actually there is a registered user named abc at system1 (192.168.0.20) having context [payasyougo] which is used to do outbound calls. we want to use this user's context and account so that when we register

I'm trying to figure out how to reload iax2 (without breaking existing calls) so it can listen on a new IP address (like "ip addr add local ..."). This alias IP is added/removed by a custom process (script) for clustering purposes. The iax.conf file contains "bindaddr=0.0.0.0". I tried a "iax2 reload" (executed without errors or warnings) but I'm still not

I just found an old bug report at bugs.digium.com with exactly the same problem. It's really too bad this bug wasn't addressed: http://bugs.digium.com/view.php?id=7315 --- On Fri, 1/9/09, Vieri <rentorbuy at yahoo.com> wrote: > I'm trying to figure out how to reload iax2 (without > breaking existing calls) so it can listen on a new IP > address (like "ip addr

Hi, I have IAX trunks between Asterisk servers. They receive calls on ISDN cards and Dial() through the IAX trunks to the "primary" Asterisk server where all the SIP phone extensions are registered. The IAX trunk settings are something like this (all servers have this identical except for the "host" field): [inbound] deny=all allow=alaw allow=gsm type=friend

Has any one used this? U(x[^arg[^...]]): x - Name of the subroutine to execute via Gosub arg - Arguments for the Gosub routine Execute via Gosub the routine <x> for the *called* channel before connecting to the calling channel. Arguments can be specified to the Gosub using '^' as a delimiter. The Gosub routine can set the variable ${GO

I am in training all day Dec 15th-16th. Please call on my mobile if there is anything that requires my urgent attention. _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com http://lists.xensource.com/xen-devel

https://bugzilla.netfilter.org/cgi-bin/bugzilla/show_bug.cgi?id=70 Summary: udp connection(snmp) not being tracked. Product: netfilter/iptables Version: patch-o-matic Platform: All OS/Version: Debian GNU/Linux Status: NEW Severity: major Priority: P2 Component: connection tracking AssignedTo:

https://bugzilla.netfilter.org/cgi-bin/bugzilla/show_bug.cgi?id=70 laforge@netfilter.org changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |RESOLVED Resolution| |INVALID Summary|udp connection(snmp) not |udp

Your email client cannot read this email. To view it online, please go here: http://obvestila.bizidan.com/display.php?M=272287&C=9501c1f306f9ce4b2d530d28eb764580&S=217&L=5&N=228 To stop receiving these emails:http://obvestila.bizidan.com/unsubscribe.php?M=272287&C=9501c1f306f9ce4b2d530d28eb764580&L=5&N=217 -------------- next part -------------- An HTML attachment was

Your email client cannot read this email. To view it online, please go here: http://obvestila.bizidan.com/display.php?M=272287&C=9501c1f306f9ce4b2d530d28eb764580&S=220&L=5&N=232 To stop receiving these emails:http://obvestila.bizidan.com/unsubscribe.php?M=272287&C=9501c1f306f9ce4b2d530d28eb764580&L=5&N=220 -------------- next part -------------- An HTML attachment was

http://bugzilla.mindrot.org/show_bug.cgi?id=70 Summary: Allow 'authorize host' questions to be able to be answered via GUI app Product: Portable OpenSSH Version: -current Platform: Other OS/Version: other Status: NEW Severity: enhancement Priority: P2 Component: ssh

Has anyone come across the Cisco ata 186 not passing *70? When I press *70, the ata just goes back to a dial tone. The strange thing is, its only *70 and not the reset of the 70's. *71, *72, etc all go through fine. I've tried removing the the dialplan all together from the ata to try and let it pass, but no go??? If I setup an IP phone to its extension, *70 goes through fine, so i know

Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAM about 75 Polycom Phones, one E1 for incoming calls. We have program a page system with the page command and the auto answer funtion of polycom. We have detect via diaplan if the phone isn't in call we place the call. All this via Macro. But in the our that they are not many calls. So much user that can be page.. The

Ok, I did check it before and nothings related to this "#" key, if it's then system will announce that "Please key in the extensions", but not in this case. By default, the blind transfer is #1..... Some one can help? >Check your features.conf file for conflicting key set. # is the default >key for blind transfer feature. >[]'s >MM chan (Alpha

Hi all, I recently purchased a Junghanns ISDNguard and to my horror I found out: - Junghanns technical support is non-existant - I can't use it without recompiling Asterisk with res_watchdog My situation: - Recompiling Asterisk with thrid party code is not an option us - We only need it for manual failover - don't require all the fancy monitoring stuff I need a simple shell script

Hi, I am having very strange situation. I have my sip peer located over the internet and I am able to connect and dial to it. the problem is that, if yesterday i was connected with it via my asterisk client and dialing to it normal way, today when i run asterisk on my client, my sip peer becomes unreachable. there is no change in settings or anything else. it may become reachable after one hour