similar to: Can be called on FreeWorldDialup/IAX channel, but can't make calls

Hi, I have 10 DID numbers. Calls coming from the PSTN network are routed correctly to the SIP users based on the number that was called. But when sip users call the PSTN network, the CallerID should be set to correspondent with their DID number. At the moment I can set the CallerID to a global number, but I have no idea how to check who's making the call. All sip users start in the context

I hadn't used FWD for quite a while. A customer sent me an email last week, "Is FWD broken when one tries to use it with IAX?" I have been playing around, and indeed seems to be the case. Is there anyone out there successfully using the two of them together? Thanks. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be

I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the

Hi, I'm installing Asterisk on Ubuntu 6.10 When I first compiled the zaptel package I used: make clean make make install So far so good, but the following command failed: make b410p I did some digging on google and found a guide on how to install it manually, but the result was the same. I got these files ftp.digium.com/pub/telephony/zaptel/b410p/misdn-b410p.tar.gz

Hi, I've posted a simular message little over a week ago so sorry for reposting. I need to register to freeworld dial up from behind a nat. Using the xten software sip client works fine but with asterisk I don't know how to do it. Last time I posted I got different responses. Some saying I can't register with an outbound proxy from asterisk others said they have done it. If it is

Hi, I've been able to make make install the Zaptel drivers (1.2). I'm using a b410p so I executed the following command make b410p. I tried this on multiple machines, but it always failes: root@asterisk:/usr/src/zaptel-1.2.11# make b410p [ -f misdn-b410p.tar.bz ] || wget ftp://ftp.digium.com/pub/zaptel/b410p/misdn-b410p.tar.gz --23:59:54--

I have configured iax.conf and extensions.conf as instructed on FWD website (http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76) and I can successfully receive calls and make test calls to 612, 613, etc. The problem is that I can not make a call to another FWD user. Here is what asterisk says: -- Executing [393xxxxxx@default:1] Set("Zap/1-1",

Hello, I have 2 channels in iax.conf [iaxfwd] type=user callerid= Free World Dialup inkeys=freeworlddialup auth=rsa context=incoming qualify=yes [iaxfwd-outbound] type=peer host=iax2.fwdnet.net username=xxxxxx secret=*********** auth=md5 The problem is: When I tell FWD to call me I have this output in my asterisk consol: Executing Dial("IAX2/iaxfwd-outbound-3",

I have a Polycom Soundpoint IP 500 that I have been using with Asterisk for a few weeks. It has been working OK, no major problems other than a freeze up every now and then, until today. The power apparently went out last night and for some reason the phone appears to be working but I keep getting the following errors repeating over and over in my Asterisk log file (IP's X'ed out): Aug

We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one phone setup as the receptionist phone, using hints to show busy office lines. This all works as expected. This is a new installation, and people are just starting to setup their phones. For those of you not familiar with SNOM phones, there is a row of keys on the right side of the phone which SNOM calls function keys. In

Hi All, I have an small issue, which is not creating any problem on working syatem but not sure about the problem that is why eager to know about it. I had installed Asterisk realtime with Asterisk 1.4.41. Every thing is working good but getting warning at Asterisk CLI. [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] WARNING[17811]:

Hi there, I'm getting a bunch of these errors from Polycom phones in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal I've searched the Wiki and archives to no avail - what do these errors mean? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

I am trying to set up the BLF on a GXP2000. Currently what I have is extensions.conf: [globals] polycom430=SIP/101 [internal] exten => 101,1,Macro(voicemail,${polycom430}) [macro-voicemail] exten => s,1,Dial(${ARG1},10,tT) exten => s,2,VoiceMail(u${MACRO_EXTEN}@default ) exten => s,102,VoiceMail(b${MACRO_EXTEN}@default) [ext-local-custom] exten => 101,hint,${polycom430}

Hello I'm using the Fail2ban. I configuration below. I want to try to prevent the continuous password. Fail2ban password that does not prevent this form. (Asterisk 1.8 / Elastix interface) What could be the problem ? Asterisk log; "Registration from '<sip:3060 at sip.x.eu;transport=UDP>' failed for 'x.x.x.x:32956' - Wrong password" Fail2ban asterisk

Hi all, I am getting the following popping up in my asterisk CLI. Everything seems to working ok, but I'm curious as to what exactly these messages mean: >>>> Dec 23 12:28:44 NOTICE[9858]: chan_sip.c:11123 handle_request_subscribe: Got SUBSCRIBE for extension 95555555555@Management from 192.168.1.104, but there is no hint for that extension <<<< Thanks for

Recently, I got the following error messages in CLI periodically. Jan 20 17:43:18 ERROR[8641]: chan_sip.c:11002 handle_request_subscribe: Got SUBSCRIBE for extension XXXXXXXXXXX@from-int from 192.168.0.123, but there is no hint for that extension I have no idea what the error message tell me. I am sure I haven't that account XXXXXXXXXXX in my database and there is no hint extensions in dial

All, I am setting up asterisk on a nslu2 (Linksys) using unslug. Everything is working great except that I have a polycom 301 and I cannot get the message indicator to work. I have created the users and mailbox in users.conf and I can manually dial the mailbox (*986000). Last thing is I am not using config files for the polycom just web browser. Can anyone point me in the right direction I

Hello Asterisk-Community, I'm having an error with my BLF configuration on my asterisk...i've configured the sip peer like this: [8250] type=friend callerid=Extensi?n 8250 <8250> canreinvite=no context=pbx9 dtmfmode=rfc2833 host=dynamic insecure=no language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100

Ok, now that I've learned I cannot use any variables when using the `hint` priority (for BLF), I figured I'd try to use the next best thing: hardcoded values using realtime. This way I avoid variables such as ${ACCOUNTCODE} but I can at least change the DB more easily than text files. This is the appropriate line in the DB:

Hello, Sorry for what may be a basic question, but I have spent a number of hours trawling the forums without resolving the problem, and hence this post. I have just started to dabble with Asterisk, as much for the learning than anything else. I created an account on FWD and used the Asterisk settings that the FWD web site recommends at