similar to: Add SIP Header for 1 SIP peer when calling a group of SIP peers

Using realtime for SIP. Using standard DB schema. Tried mailbox as varchar(50) and bigint(10) "sip show peer XXX" shows "Mailbox: " (empty) So MWI isn't working This happened before when we tried to go up to 1.8, so we stayed at 1.4 We're forced to go to 13 now. Obviously a lack of MWI is a big issue for 500 units. Any idea why we're not getting it? We're

Hey Guys, I'm after a way to monitor our sip trunks (peers) and send an email if they go down. I know I could use 'asterisk -rx "sip show peers"' in a shell script but that seems messy, especially since I'd like to monitor it fairly closely (so I'd like to run it every 20 or 30 seconds or so). Is there a better way to do it?

Hi all, I've been working on this for days and can't find a solution. I need to use AddQueueMember for my agent logins to my Queues -- but a number of my agents are outside the main server, which is connected to my asterisk network over IAX2. I can't just do a AddQueueMember(queuename) because it puts in a complicated member calleridnum like: IAX2/peername:65723/23 Which won't

Hi Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get the 'no matching peer' error when we get a dictionary SIP attack. Now the logs always show a 'wrong password' when there actually isn't a matching peer. We even have alwaysauthreject = yes in our sip.conf. Has anyone else noticed this phenomenon? Thanks in Advance Ish -- Ishfaq Malik

Hi, Is it possible to setup SIP peers with Asterisk Realtime from multiple databases? Thanks in advance. Ash

Dear all, I have Asterisk 1.4.13 as a SIP server for my company with 100 peers (I mean users) and everything work fine. I have the following question: what is the maximum number of peers that I can reach with Asterisk ??? I know Asterisk is not a SIP server basically like OpenSER, so I'm confused. Thanks a lot, Alejandro

Dear Gluster, I'm running Gluster 3.3 on a four host setup. (Two bricks per host.) I'm attempting to use this as an rsnapshot backup system. Periodically, the rsync's seem to fail, and I think this is due to underlying gluster failures. I notice this in /var/log/messages: # Jul 19 23:47:15 annex1 GlusterFS[26815]: [2012-07-19 23:47:15.424909] C

hi all! I have the following situation: 1 -------- 2 ? ? ? ? 3----------4 ? ? ? ? 5----------6 where 1 ... 6 are nodes and every direct neighbor is specified as a dundi peer (in *). When I start a dundi request, every queried node is mentioned in the dpdiscover. For example 1 sends a discover to 2 and 3, so 2 sees in the EID or

Running 1.6.2.8 and using teliax and junction for sip providers. We also have an internal sip network for extensions. On boot both providers are shown as unspecified and UNKNOWN. The internal sip extensions are found. sip reload always finds the providers. Why doesn't asterisk find the providers when it starts up? And if it doesn't find them, how long before it tries again? sean

Can asterisk log the registration date/time in a database? Is there a standard option to do this? I know it being logged in the asterisks 'full' (debug) log and we are probably able to script something with the API interface but there might be somewhat easier if there is a option to make asterisk log this information directly into a database. Thanks in advance, Bram

anybody use them on Gluster? They seem to be almost the same cost as spinning metal these days. In fact I was trying to get some 2.5 inch 2TB drives on a vendor and all they had was the firecuda SSHDs or the really expensive "Enterprise" variety. Our use case would be for VM hosting (Rep2 + Arb). I'm not sure how the SSD cache would pan out with the shards. I've googled

Hi, As you may have noticed, I've been playing around with a couple of ways of displaying peer statistics in Windows. There were previously 4 programs to do that. I have now combined them into 1 uberprogram (392Kb uncompressed). You can download the beta from: http://www.sineapps.com/down/AstWinPeers.zip or you can view a screenshot at: http://www.sineapps.com/images/AstWinPeersBig.jpg

I am about to start a program that will be generaging sip device configurations for sip.conf. My current sip.conf contains friend entries for each SIP device connected to asterisk. Should I even be attempting to split these in to seperate user/peer devices? Is there(should there be) a convention for referring to the two distinct modes of a device? Can two entries with the same name exist at

Hi to all, following the suggestion of Matteo Brancaleoni to solve my problem of anonymous call in CDRS i implemented the accountcode in any peer that i have in the sip.conf so i can use the SetCallerID to accountcode to identify the caller. When i make the sip show users the accountcode is correctly displayed but when i try to access the variable ${ACCOUNTCODE} from the extensions.conf to

Hi, notice that i have Grandstream phones and i have the problem if i activate the Send Anonymous function on them. If i do not activate that option the ACCOUNTCODE is correctly populated. SO i think it may be a bug of asterisk. I'm using Asterisk CVS-HEAD-10/07/04-18:07:25 . Thanks, Bye, Marcello

Hi After going from AST_DATA (RES_DATA) to realtime with mysql-driver my console is jam'ed with SIP Seeding peers from Astdb '000b8201XXXX' at 000b8201XXXX@81.146.XX.XX:35273 for 120 I got arround 4000 sipclients registered at that server and all the sip-client re-register every 120 sec. so the console is totaly fill'ed with SIP Seeding messages. Is it posible to not see

Hey Everyone, This is not an operational issue, and to my knowledge only effects the look of the command, but when I issue a "sip reload" then a "sip show peers" I see all of the actual usernames I have assigned in my sip.conf. However, five minutes later I reissue the sip show peers and all of the usernames have disappeared and are replaced by the sip ID. The only way to get

Good afteroon i had found a special issue while using "sip show peers" sometimes i get 6115/6115 xx.xx.xx.xx 6109/6109 xx.xx.xx.xx 6001/6001 xx.xx.xx.xx 6107/6107 xx.xx.xx.xx 00/500 xx.xx.xx.xx 7000/7000 xx.xx.xx.xx in fact it must be 6115/6115

A group which my school is part of wants to start using DNS SRV records to allow "email-style" dialing amongst members of the group. I have gotten the records in our zonefiles, and things work pretty much just fine. However, since the DNS server can only specify a host and port, there doesn't seem to be any way to authenticate the user coming in. Is that the case? Is there a

hi all, is there any way to force all sip peers to re-authenticate themselves? thanks, Paradise Dove