similar to: IAX <> IAX trunking... DP cache?

Hello, i have a simple but annoying problem. I have the following entry in /etc/asterisk/externsions.conf file: ---<Cut Here>--- exten => 10100,1,Wait(4) exten => 10100,2,Playback(transfer,noanswer) exten => 10100,3,Dial(${PHONE30},30,t) exten => 10100,4,Background(extension) exten => 10100,5,Background(is-curntly-unavail) exten => 10100,6,Voicemail(9999) exten =>

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

Does anyone have a working example of how to use the "switch" directive to peer two Asterisk PBXes? -- - Ian C. Blenke <icblenke@nks.net> (This message bound by the following: http://www.nks.net/email_disclaimer.html)

I have an Asterisk 0.4.0 install working with two grandstream budgetone 100 phones, gnophone, and kphone. This is a private network segment (172.17.x.x), with the PBX configured on my outbound firewall which has a public address (66.x.x.x). - I can make calls between phones - all extensions are working. - I can make IAX calls to IAXTEL. No problems (apparently gsm only) - I can call SIP phone

Since firmware 1.0.3.81, unless I'm imagining things, Voicemail2 seems to be having problems. The Grandstream and sip.conf were set to RFC2833 now with that setting I get extra digits during "Mailbox" and "Password" phases. 222001 instead of 2201 for example. When both are changed to "SIP info" there is no problem. But what is the new setting "DTMF Payload

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

I just had the same exact problem this morning. The only thing I've done in the last couple of days is update update zaptel. I rolled back my zaptel to 2/11/04 from 3/8/04. And kept my libpri from 3/8/04. I never had this error before updated. I had other issues, but not this one. -sb -----Original Message----- From: asterisk-users-admin@lists.digium.com

On the Granstream 102 box that I have in front of me, there is a "feature list" on the side. One of the features has grabbed my attention: " - optional voice encryption (model 102D)" Now, digging through Grandstream's site, I see that it's not offered quite yet. However, sending mail to their standard "information" email address has resulted in no

i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. trunking will save some bytes in flight iff one has four or more streams moving between two

Hi, my setup is quite simple: an asterix CVS of 2003-11-15 on a 2.4.21-debian-5 GNU/Linux box in an internal network (192.168.1.0/24, asterisk is 192.168.1.10). - with a SIP phone configured as 192.168.1.190, and with its SIP server being 192.168.1.190 - with an ISDN AVM c4 i4l card on an ISDN connection with 2 channels. I try to: - dial-in from ISDN, then transfer to the SIP

I want to create a new array which selects values from an original array based on a function of the indices. That is: I want to create a new matrix Vnew[i,j,k]=Vold[i,j,ks] where ks is a function of the index elements i,j,k. I want to do this WITHOUT a loop. Call the function "ksfunction", and the array dimensions nis,njs,nks. I can do this using a loop as follows: # Loop version:

I updated to the latest CVS tonight and now DTMF detection does not appear to work on my Cisco 7960 sip phones (can't check voice mail etc). The asterisk console is displaying these messages over and over again any time a DTMF tone is sent: NOTICE[15376]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec 101 received Downgraded to a known working CVS of about three weeks ago, and

Hi, I use asterisk with a Digium E1 (wct1xxp). On my old server, everything went fine, but after having built the card to a new one, I only have problems: -- Executing Dial("IAX2[guest@217.X.X.X:18308]/1", "Zap/1/853") in new stack May 15 14:10:37 NOTICE[15376]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time

I?ve been looking at an issue when compiling Samba on FreeBSD 13 with the vfs_zfsacl module. It seems the logic that detect O_PATH support in Samba doesn?t work right with FreeBSD 13 (since it has implemented O_PATH support) and especially with the acl_get_fd() function. This is a bug (or rather a missing feature) in FreeBSD that there is a patch for now (dunno when it?ll appear in a release

I''m creating a new domU using ''virsh create <xmlfile>'' without specifying the MAC address within the XML file, under the assumption that the dom0 or hypervisor generates the MAC address for me. Next I''d like to get the MAC address that was assigned to this new domU, and I don''t see a way to get that information from the command line via virsh

I have followed a combination of instructions from this post and have been able to get an Ubuntu paravirtualized instance running under snv_96. Is there a way to "list" the fact that it is paravirtualized? For example, xm list, virsh list, etc. thanks -- This message posted from opensolaris.org

On Tue, May 30, 2023 at 09:57:37AM +0200, Peter Eriksson via samba wrote: >I?ve been looking at an issue when compiling Samba on FreeBSD 13 with the vfs_zfsacl module. > >It seems the logic that detect O_PATH support in Samba doesn?t work right with FreeBSD 13 (since it has implemented O_PATH support) and especially with the acl_get_fd() function. This is a bug (or rather a missing

On Zed''s suggestion, I caught two new spinning mongrels and sent a SIGUSR2. The code appears to be stuck in read_multipart for both processes: # kill -USR2 6109 ** USR2 signal received. Thu Sep 21 14:55:39 EDT 2006: Reaping 1 threads for slow workers because of ''shutdown'' Thread #<Thread:0x419d7ce0 run> is too old, killing. Waiting for 1 requests to finish, could

This is important so please read this message very carefully. There is a DoS for Ruby''s cgi.rb that is easily exploitable. The attack involves sending a malformed multipart MIME body in an HTTP request. The full explanation of the attack as well as how to fix it RIGHT NOW is given below. Most of the work was done by Jeremy Kemper and Jamis Buck. They did all the work of building the

First off: Our clusters are LVS balanced Apache 2.2.3 + mod_proxy_balancer + gem mongrel 0.3.13.3 / mongrel_cluster 0.2 + memcached / gem memcache_client + gem rails 1.1.6 on debian boxen, and a pgcluster backend. On 2 of our deployed clusters, we are getting the "spinning mongrel" problem. As the clusters are very low volume right now, it takes days to collect a spinner, making it