similar to: Ser + Asterisk & DMZ

I'm using asterisk to forward some sip incoming calls to ser, I've noticed that every call * passes to ser has sip:asterisk@IPADDRESS as header. Is there a way to make * pass the number of the caller in sip address to ser. I mean if I get a call from PSTN number 123456, how can I pass sip:123456@IPADDRESS to ser? In docs I found fromuser, username, callerid options, but they affect

You will also need your SIP clients that are behind the same NAT to support ICE (Interactive Connectivty Establishment) if you want calls between them. Xten Eyebeam and Snom phones are the only ones I'm aware of that support it. On 7/5/05, Ricardo Martinez <rmartinez@redvoiss.net> wrote: > And even worst. > There are some kind of NAT that STUN does not work. > You can check

hello, i'm reading and trying to configuring my mailserver following this howto on the dovecot wiki: http://wiki.dovecot.org/moin.cgi/DovecotPostgresql but i got always this problem: ERROR: permesso negato per la relazione postfix_virtual in english is "permission denied for the relation postfix_virtual". I dont know where the problem is, but if i send an email from localhost

hello, i've implemented this on my mailserver : http://wiki.dovecot.org/moin.cgi/DovecotPostgresql but now mailman is broken. This is the error i got from postfix: Recipient address rejected: User unknown in local recipient table (in reply to RCPT TO command) of course the user mylist at mydomain.org does not exsist on the postgres db, but how i can fix this issue? someone wanna help,

Ok that is pretty much what I do.. Except.. Most of my net sends don't end up working.. They fail unless I explicitly do a nmblookup and specify the broadcast to find that computer.. Seeing how im dealing with the entire contents of 10.x.x.x around 20,000 computers that I may have to contact for some reason.. Its easy for me to know there netbios name.. As its there login name.. However

Hi I have a cisco 7960 Phone that connects to my Asterisk server without a problem. But when I call the echotest it just hangs up, echotests from other VoIP providers works just fine. I have tried a softphone and it works just fine. The error I get when the 7960 calls is this: -- Executing Playback("SIP/2000-180c", "demo-echotest") in new stack -- Playing

Hi, I tried to play the standard demo-echotest file !. It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error: Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A Sep 8 14:58:33 WARNING[-182461520]: file.c:779 ast_streamfile:

After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very strange problem. There is no sound with Playback() or Background() commands. Even though, Asterisk console shows the file is being played when I call the extension (i.e. echo test), I can't hear anything. My echo test extension looks like this: exten => 600,1,Answer exten => 600,2,Playback(demo-echotest) exten

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and

Hello, 1) Thanks - shorewall save me a lot of time! 2) I try - exactly: I must :-) - configure a VPN server behind 2 NATs. My situation: RoadWarior - INet - ISP Router (NAT+PortForwarding) - Inetranal Router (running Shorewal, NAT+PortForwarding) - Inetranl VPN Server If RoadWariror try to connect Internal VPN Server then connection failed with "GRE: Bad check chcksum from pppd"

Hi everybody, I would like to count how many times names in list L, nombreL, apear in list C, nombreC. Can I improve the next program? cuenta <- 0 topL <- length(nombreL) topC <- length(nombreC) for (i in 1:topL) { for (j in 1:topC) { k <- grep(noquote(nombreL[i]),nombreC[j])

Hi to everybody, I have the problem that nearly no information are displayed on the Asterisk CLI (asterisk -r). In former times (before patching Asterisk 1.0.2 with ast_data 1.0.2) it looks e.g. like this: --- snip --- -- Registered '96' (AUTHENTICATED) at 212.202.169.118:4569 -- Accepting AUTHENTICATED call from 212.202.169.118, requested format = 1024, actual format = 1024

New install, Calls are working phone to phone using gsm, ulaw or alaw codec but when try and echo test or voicemail there is no playback. I've tried turning on and off every codec and still no luck. Asterisk says it's playing the sound file but I just don't hear anything. I can't find any reason for this. I've tried the latest tar and CVS with the same result.

Hi all, Couldn't find a place to search the list archives... I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. I went by the instructions pointed at the "10 minute guide", located here: http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart. I installed it on a Slackware 10.1, by using no more than "make

Hi all, Couldn't find a place to search the list archives... I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. I went by the instructions pointed at the "10 minute guide", located here: http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart. I installed it on a Slackware 10.1, by using no more than "make

I've got an SPA-841 SIP hardphone connecting to my asterisk server across the internet through a couple of NAT routers. Everything works great (I can initiate calls, receive calls, hear audio both ways, etc.) except for one thing. When I hang up the phone, the connection in asterisk doesn't disconnect. The phone is idle and things everything is fine, but Asterisk still show an open

> Olle E Johansson wrote: >> >> 23 apr 2007 kl. 19.55 skrev Russell Bryant: >> >>> John Todd wrote: >>>> To morph this into a -dev thread: if this patch were to become (again) >>>> useful and error-free, is there any objection or usefulness in adding it >>>> to TRUNK? Personally, I think there is, if there is a method by which

Hi list I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk solutions and appliances. I installed TrixBox on a litle PC @ home and a x100p card which is recognized as a Zaptel card, I made some in/outbound routes and they seem to work but I have a problem with SIP softphones. I created 2 estensions 1000 and 1001 they're both in different cities, when I 1000

Hi all, running beta 5 doesn't allow ssl connections: SASL authentication failed Same config runs well with beta3. I tried as well latest CVS, but this gives me the same error. Verbose debug output: dovecot: 2006-04-09 17:17:51 Info: auth(default): client in: AUTH 1 PLAIN service=IMAP secured lip=192.168.200.1 rip=192.168.200.202 dovecot: 2006-04-09 17:17:51 Info: