similar to: Passing SIP Subscriptions???

I'm pretty sure I already know the answer to this, but... Is there a way to copy/transfer/replicate sip subscriptions from one asterisk system to another, for the purposes of HA? You coudln't even write a script to do it I don't think. You can do an 'asterisk -rx sip show subscriptions' but there'd be no way to repopulate it on a second system. Yes/No? Doug.

It seems that when you use Realtime static and possibly realtime realtime for sip users, that Asterisk fails to create the regexten context for DUNDi. Someone else had the same problem back in July. Doesn't look like they ever had a resolution. <http://lists.digium.com/pipermail/asterisk-users/2006-July/160105.html>

Before I play around with this again in 1.2.1, regexten is still essentially broken, correct? The misconception seems to be that it allows you to execute a command upon registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is for. After reading the developer discussion though, it definitely seems to be broken. Is it fixed yet? Doug.

All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk.

Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate

Somewhat off topic. We upgraded a Polycom phone from SIP v1.6.3 to v1.6.6 The phone will no longer send SIP subscription messages for buddies to Asterisk. I have broken the directory file down to make it as simple as possible. Here is what it contains. <?xml version="1.0" encoding="UTF-8" standalone="yes"?> <!-- $Revision: 1.2 $ $Date: 2004/12/21 18:28:05

On Fri, 2006-12-08 at 04:26 -0700, Douglas Garstang wrote: > Hi Steve. > > Thanks, but unfortunately, I can't be involved in that. We are > running Asterisk in a production environment and we're using > 1.2, not 1.4. I don't have the resources to work with 1.4. > Last time I filed a bug against 1.2 I got told off. >

Off topic. Anyone know where I can get Polycom SIP software v1.6.6, unofficially? Not that Polycom is analy retentive, or anything like that... Doug

Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want

Well, this is weird. After receiving a sip subscribe message from peer 2529266, here's what Asterisk responds with: -- (14 headers 0 lines)--- Found user '2529266' Looking for 2943110 in bell_CallStart (domain ua2.ipt.xxx.com) Dec 26 10:19:34 NOTICE[27345]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'bell_CallStart' Transmitting (no NAT) to

Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and

Great Email. I'm going to respond to some of the points. "Q: What are the plans for HA? A: With a configuration using DNS-SRV and DUNDi, you can create a pretty resiliant setup now." That's BS. Last time I checked, Asterisk's support of SRV was to only grab the first SRV entry. Period. If it doesn't try any more SRV hosts after the first fails, just exactly how

Is anyone using realtime sip for friends (ie phones) with multiple Asterisk boxes all pointing to the one central MySQL database? Does it work? Are phones that are registered to the database from Asterisk box able to reach phones registered to the database from another Asterisk box? Doug.

I'd love to understand what the function of the peer list returned by 'sip show peers' is for, especially when Realtime is used. If I start Asterisk with realtime enabled, a 'sip show peers' yields none. As each peer (phone) registers, or a call is made to the peer, Asterisk adds them to the list returned by 'sip show peers'. Correct? Apparently Asterisk doesn't

Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' soft-key to work? When you change the status in this way, the phone does not send any communication to Asterisk, and it seems to have no effect in incoming calls. So... what's it for? Doug

Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug.

Can someone tell me why this is not valid... [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@Local) exten => 2000,1,Macro(DialProxy,115551212) exten => 3000,1,Queue(testq||||45) while this is: [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@start) exten =>

Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says "the context for the voicemail box that we're looking for in the dialplan for the jump to the

I recollect that there used to be a fixed, finite limit to the number of call groups that could exist. Does anyone know if that limitation still exists in 1.2.1, or maybe where I could look in the code to find out if it's a fixed length array or not? Thanks. Doug.

Does anyone know if/how I can get Asterisk to listen for SIP/RTP/IAX/whatever on multiple network interfaces? Thanks, Doug