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I have a scenario here with IP phones, on a private 192.168 network connecting to an Asterisk box, also on the same 192.168 private network. We'd like to have the Asterisk box also be able to send traffic to the public IP space. For this, we would need to multi-home the box, and put two network cards in it, with two IP addresses, one on each network. I know from past experience that

I don't know if this is possible, and I can't quite get my head around how to do it... If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: +---+ +---+ | A |-----| B | /+---+ +---+\ / \ Phone1 Phone2 Is there a way configure re-invites

We're dialing a disconnected number via Level 3's vector network, and are receiving this. The response has SDP in it. Apparently, Level 3 is playing early media. Asterisk doesn't seem to know what to do with SDP in a 180 RINGING, and just plays ringing. What am I missing here? How can Asterisk see there's SDP, early media, in the response and act accordingly? SIP/2.0 180

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

I'm trying to find a high port count ATA device. We have a lot (up to 110) analog devices that we need to bridge to IP. Rather than go out and buy 110 ATA's, it would make more sense to buy a single chassis type device with some number of ports and blades. Anyone know if such a device exists? Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed...

Has anyone tried to compile the current version of MySQLPool from http://www.yosd.at <http://www.yosd.at/> against Asterisk 1.4.4? It seems to not compile... [root@itsentinel res_mysqlpool]# make gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -I/usr/local/mysql/include/mysql -D_REENTRANT -D_GNU_SOURCE -O6

All, I am trying to find a SIP ITSP that honors dial around compensation. We are adding a Flex ANI code to our outgoing SIP invites by appending an isup-oli tag to our From: address, like this: INVITE sip:18889996563@carriers.icall.net SIP/2.0 Via: SIP/2.0/UDP xxx.y.34.201:5060;branch=z9hG4bK7f314484;rport From: "Dougs Payphone"

Where's Steve Murphy when you need him? :-) This doesn't seem to work in AEL2... Macro foo(arg1) { ..... Includes { Hangup; } } The error is: File: /etc/asterisk/extensions.ael, Line 59, Cols: 5-12: Error: syntax error, unexpected KW_INCLUDES, expecting '}' The same error does not occur when the includes is in a context. I need to

Ok, I could have sworn this was fixed in Asterisk 1.2, but it seems in Asterisk 1.4.4, that doing a reload, or even an 'extensions reload' will clear any extensions that have been created by regexten. This is VERY bad! Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can anyone recommend any wholesale SIP termination providers that will automatically charge a credit card? Most seem to want upfront payment and a credit balance but that sucks when you have to watch it like a hawk to make sure the balance never hits zero. I'm looking for a provider that can automatically charge a credit card. Douglas. -------------- next part -------------- An HTML

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

Does anyone know how the Linksys PAP2T ATA's can be mass provisioned? Documentation seems to be sketchy, even on the Linksys web site. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070607/3f90695c/attachment.htm

How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070803/c6d473ce/attachment.htm