similar to: Extensions SIP problems.

I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial("IAX2/firefly@89280250/3",

Ok I have spent the last week working on getting my small PBX to work. I will in the end only have 4 SIP extensions being either softphones of IP phones. Currently only 1 SIP config for testing. And at the this point it should be all fairly easy with all inbound and outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via IAX. Inbound does work in it's current basic state.

Hi Muhammad From: "Muhammad Talha" <talha@worldcall.net.pk> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, December 22, 2004 2:07 PM Subject: Re: [Asterisk-Users] Zaphfc/BRI Configuration help > Thanks for lan for your reply can you share your extention.conf > setting . > >

Hi, I was messing around with FireFly last night and got asterisk to crash hard. It looks like the bug is a division by zero in chan_iax2.c. I reproduced it and here are some infos I got from gdb: [Switching to Thread 245775 (LWP 23251)] 0x41154918 in calc_timestamp (p=0x816b710, ts=0, f=0x424eef24) at chan_iax2.c:2896 2896 int diff = ms % (f->samples / 8);

Ok I have a challange that I can't seem to find a way to fix it. My Voicemail in * timesout after 30secs without fail everytime no matter what I do. I have incomming calls comming in through Freshtel IAX2, if it goes to SIP extension when it is online it can hang on for what ever time the call goes for. If however it goes to the Voicemail it will timeout at 30sec and I can't seem to

As my previous try on getting an answer was hijacked I thought I would try again. Ok I have a challange that I can't seem to find a way to fix it. My Voicemail in * timesout after 30secs without fail everytime no matter what I do. I have incomming calls comming in through Freshtel IAX2, if it goes to SIP extension when it is online it can hang on for what ever time the call goes for. If

Hello, I installed 1.2.5 and realtime SIP. The connection to the DB is OK because I can get the values from the CLI. Here are my 3 different cases: 1- If I put an unexisting user, I get 404 and I am not able to dial. 2- If I check "Disable registration" within Firefly it does not register but I am able to dial a destination (...) 3- If I leave registration ON, I get the 404 message

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

If I have an IAX client (Firefly softphone in this example), and the client is not registered at the moment because they are not connected to the network and someone dial that extension, they get the user's "I'm on the phone at the moment" message vs. the "I'm unavailable" message. Is this by design? Here's the extension in question's dialplan:

Recently I've noticed two bits of odd behavior with respect to regcontext/regexten in CVS HEAD & 1.2 Beta1, and I was wondering if anyone could shed some light on this. I've set up a regcontext in sip.conf. I've set up two users with regexten entries, one in sip.conf and one in a mysql realtime table. The first bit of oddness is that regexten seems to work somewhat as described

Dynamic DNS updating is failing (which is bizarre, because I could have sworn I'd had it working before). Help? Setup: Samba 4 DC running bind 9.9.2, Samba 3.6.3 member The output of "net -d10 ads join" is attached, compressed. Interesting portions of named.conf: options { (no allow-updates section) ... tkey-gssapi-keytab "/var/lib/samba/private/dns.keytab";

Hi everyone! I've set up asterisk ip-pbx to implement IVR menu and encountered such a problem: when users dial the destinaion phone number and end it up with "#" asterisk still waits until timeout in WaitExten() is reached. // Here comes the context where user is prompted for a dest. number: context ivr-dialout { s => { Background(enter-your-dest-number);

Hi, I am running Redhat RHEL 4, authentification is via kerberos against and AD server, usernames are supplied via ldap service running on another redhat box - winbind has been seg faulting repeating when accessing samba - always the same error message... see logs below - can anyone tell me whats going on? Mar 14 16:12:45 firefly winbindd[14752]: [2008/03/14 16:12:45, 0]

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work

Hello, On a computer, a sparc64 with Debian, with hostname tw89 and domain name KUDDE thinks `smbpasswd` that the domain name is TW89 | tw89:~ | # grep workgroup /etc/samba/smb.conf | # Change this to the workgroup/NT-domain name your Samba server will part of | workgroup = KUDDE | tw89:~ | # ldapsearch -LLxW -H ldaps://tw89 -D cn=admin,ou=people,dc=gst,dc=stappers,dc=nl \ | -b

Every so often, and it seems that it happens only when a call is in progress, all 24 Zap channels get reset. All channels are opened and then timeout. This causes the in-progress calls to terminate. There are no corresponding Red/Yellow alarms on wither the PBX or Asterisk although we do receive a fair amount of Blue Alarms. The Asterisk server is connected to a legacy PBX through a Digium

I did get it to work, and can place and receive calls through the Firefly network via *. Compared to iaxtel or FWD, there is a significantly higher amount of latency, but it is workable. For some reason, this needed to be the last entry in my iax.conf or it would try to authenticate with a different user ID when receiving calls (and obviously would fail. Relevant section from my iax.conf:

I found version 2.0.0 of Firefly on the Freshtel site, but it only has the network setup options for the Freshtel network, despite the final statement on the page http://www.freshtel.net/firefly/download/ that says: ----------------- Standalone SIP / IAX mode: If you want to use Firefly on our network (with your own voicemail etc.) you will need to register a Firefly number. However, you can

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

So far, I have three machines in the domain: kaylee -- the DC. Gentoo. samba 4.0.3 saffron -- client. Gentoo. Samba 3.6.12 wash -- client (Also network router). Debian. Samba 3.5.6 I'm using bind_dlz as a backend, for the record. I've joined saffron to the domain successfully, and the record shows up in DNS. $ samba-tool dns query kaylee firefly.michael.mol.name saffron all Name=,