similar to: Voicemail after one ring?

I'd like to be able to add additional fields to the the Asterisk database. I'm using Mysql for most of my data lookup and manipulation, and it seems to work pretty well. In keeping with what I know how to do, it would be very handy to be able to insert say a "call forward number" into a customer record. That way, I could automatically route calls to extensions to a forwarded

Hey list (again - annoying bastard I am) I've played with Firefly/* for a while and I have yet to find a way to have * send voicemail notification to Firefly. It appears possible using SIP (no clue whether Firefly supports it) in the sip.conf file, but there's no mention of anything voicemail-related in the IAX.conf file. I'm using IAX with Firefly, so that might just be the

I'm having trouble with call pickups. Suppose ring group is 100 and has extensions 101 and 102. Someone calls 100, 101 rings and 102 wants to pick the call up. If 102 dials **100, call pickup works. If 102 dials **101, call pickup fails. In my dialplan I have: exten => **101,1,NoOp(pickup extension) exten => **101,n,Pickup(101) exten => **101,n,NoOp(pickup group) exten =>

Even better you can set your firefly softphone to auto answer so that you don't even need to be near the pc to answer. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Linn Boyd Sent: Friday, March 04, 2005 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Is it possible to set up Asterisk without any of the cards? I'm interested in setting it up for the company I work for, but I would like to set it up and see how difficult it will be before I start having the company spend a chunk on equipment. Additionally, what phones can be used with Asterisk? we currently use a NEC Nitsuko phone system with phones, but I have been confused as how to set

Hi, I've successfully used the 'd' flag in Dial() so that when I dial into my phone system from out there in the PSTN network I can press the 2 key while the phone is ringing to listen to my voicemail. It seems that one issue is that the public providers do not deliver DTMF, or anything, until the phone is answered. This is for security reasons and sounds like a good idea to me.

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

I have just got my X100p card and am trying to install it. I have built and installed the zaptel drivers without any apparent errors and I am now trying to get them working. I have the following in config files In /etc/zaptel.conf fxsks=1 In /etc/asterisk/zapata.conf signalling=fxs_ks context=default channel => 1 Now when I try ti modprobe the drivers this is what happens: Command:

Asterisk Users: We have our * dial plan set up to ring 5 phones in the office and it delivers the call to the first that picks up their receiver. The problem is that after the pickup, everyone else's SIP phone keeps ringing at least once and sometimes twice. This interferes with the conversation, while others pick up the phone and get nothing. Does anyone else have similar problems or

Hi, all I am trying to set up voicemail. I've done it to the point where I can leave messages. How do I retrieve them? Actually I have few questions: 1. I want voice mail to be available at certain extension, say 100. How do I set it up so all users can call this number and get to their respective mailboxes. 2. How do I let users to create their own voicemail passwords from the phone? 3.

Hi, when I press reject on Firefly, another end hears congestion, but I'm still listening to ringing... Also how you determine which version of Firefly you have? I also don't understand licence completely and would like to get this answer - can I distribute Firefly to commercial users if I don't charge anything ? Regards, Rob.

Hello Maybe a silly question, but after some searching couldn't find answer. Is there a number I can dial to pickup and listen to my voicemail messages on my SIP phone? I am used to eg dialling *17 to pickup my voicemail messages on Avaya system? Angus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Everyone, I've got a weird problem with both Firefly & iaxLite (both IAX softphones). They don't seem to stop ringing when an incoming call is make to them. If the call is answered the conversation starts both ways but the ringing sound still keeps going and the softphones keep displaying that a call is coming in (but they do not display that the call is answered). I read

Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device

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, 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]

Hello, a user outside the office regularly gets a call from ext. 101 but that extension does not exist in my extensions.conf. when the user pickup the phone no one answers. Any Idea how to fix this issue? that user uses Polycom SP 450, Thanks in advance, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

1. We just dial the extension directly and have speed dials setup for the first 6 parked positions. We don't use *8 at all. 2. Change the config on the phones under Account to "Send DTMF via RTP (RFC2833)" -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Noc Phibee Sent: Thursday, February 08,

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:

Hello List: I'm working on a funny scenario, where I'm bouncing calls from a Cisco call center into asterisk. Cisco call center has some logic that if a customer calls in, an agent is logged into a given extension... if Cisco sends a customer call to that extension, and there is a ring with no answer after a preset amount of time, Cisco concludes the agent is unavailable, kicks the agent