Re: [asterisk-users] call takeover?
Hi C., Check out the "pickupgroup" and "callgroup" options in sip.conf -- these should accomplish what you're looking for: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf More about this feature is defined here: http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups If you need to be more specific in what to pickup, you could likely use the Asterisk Manager API's "Redirect" action to redirect the call to another device: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect Hope that helps -samy Csibra Gergo wrote: Hi, situation is the following: There's an inbound call, that rings on SIP/tel21 (ATA is PAP2). At the time, bobody there, but a lazy people sits by SIP/tel22 (about 5m distance) and he want to takeover the call. How can I do this whit asterisk? Ok. I can do with call parking, but with call parking on SIP/tel21 must I call the parking extension too, and if nobody picks up the phone, the fax machine (the SIP/tel21) must answer it, and the fax machine can not call the parking extension. ps.: sorry for starting new thread with reply, but I can not send mails to this list otherwise. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding NAT Traversal
Hi guys, In addition to the required RTP ports needing to be opened on a NAT router, the primary difference between HTTP and SIP is that SIP opens up an additional session (the RTP session) on a completely different port (and possibly IP) as stated below, and even more specifically, the SIP packet defines where to connect to. When you send an HTTP request, the only place your IP address is located is in the IP header itself, not the HTTP header of the packet. A typical (see: symmetric) NAT changes your internal IP address in the IP header of the HTTP you've sent to the IP address of the gateway, sends it along its merry way, and waits for it to come back (with the same source and destination ports/IPs, but swapped). Once it comes back, it contains all the data you asked for. At no time does the remote HTTP server ever have any idea that you're on an internal IP. SIP, on the other hand, typically only handles the call setup/etc, but not the audio (RTP) stream. When you send a SIP INVITE, included is an SDP header which contains _your_ IP address in addition to the IP header of the packet. While your router modifies your internal IP address to the gateway address in the IP header, it does not alter the internal IP address in the SDP header (unless it's a SIP-enabled router/fw). The destination then attempts to connect to the RTP IP+port you provided, which is an internal IP address and probably not accessible from wherever you're sending your INVITE to. Asterisk's sip.conf's `externip` and `localnet` lines are used to resolve this when making calls through Asterisk. The way SIP works is much more like active FTP connection rather than HTTP, where there are actually multiple connections and the important data you're trying to transmit (audio, data, pron) is sent on a different port. One of the benefits is that you don't need to transmit the data yourself if you're simply handling the session data between two remote parties. Being a SIP proxy in this scenario is very similar to FTP's "FXP". Hope that helps -samy Mojo with Horan & Company, LLC wrote: I'll take a stab at the first one, but I am probably gonna get nailed for my own rudimentary understanding of it... hugolivude wrote: I understand how sitting behind a NAT could cause problems for a SIP UA. The SIP UA would create SIP mesages using IP addresses from inside the network (i.e. 192.#.#.# or 10.#.#.#) and these IP addresses are of course unnavigable for the recipient. What I don't get is why don't web browsers suffer the same problem? When the initial http request goes out the packet has a return address with a source port on the (for example) 192.x.x.x machine. the NAT router does the same thing, sending it out into the world, with a return address containing a source port on the NAT router. when it receives a response, it knows which internal box to route it to, things are OK. The HTTP server simply sends its reply to that return address, and it's routed back the way it came. I think the biggest problem with SIP is the RTP ports. The initial SIP request goes out (for example) to port 5060, and FROM port 5060 as well. The response needs to get back to the SIP UA on that port (which would limit the nat router to only be able to deal with ONE internal ua at a time, if they were both using the standard port 5060), which could conceivably happen easily enough. But in the SIP "handshake" more ports are chosen, typically in the 10,000 to 20,000 range. The NAT router would then need to be configured to direct that anticipated RTP stream to the proper internal client. That just doesn't happen :) For various reasons, I'm not too partial to UPnP, but maybe there needs to be a SIP UA that uses UPnP to configure a NAT router for it, when an RTP stream is begun? Now the clincher to all of this is that I'm merely talking about the ip packets transferred and their return addresses. While I'm not qualified or experienced enough to comment on problems that might arise with the contents of the SIP headers themselves, I suspect that's where the REAL trouble lies with SIP NAT traversal. The SIP UA needs to put the proper return address in the SIP headers before the lower layers of the OSI model take over. It can't know its externally-visible ip address unless A) the user manually enters it or B) it can ask some outside server what it's perceived address is. Moj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared
Correct -- any changes to just zapata.conf do require a full restart of asterisk (but do not require reloading of zaptel modules) Rich Adamson wrote: I'm 98% sure the zapata.conf changes require a full stop/start of asterisk. Are changes to the zapata.conf file read on the fly or do you have to restart the asterisk process? On 8/29/05, Matt Fredrickson <[EMAIL PROTECTED]> wrote: On Sun, Aug 28, 2005 at 02:52:20PM -0400, Andrew Kohlsmith wrote: On Sunday 28 August 2005 11:59, Steve Underwood wrote: I don't follow why knowing that impedance mismatch is the problem has stopped you making fxotune fix it. :-\ Where you the one who asked me how to make fxotune work well on IRC? Someone asked a while ago, and said they were working on a faster tuning algorithm for fxotune. I've forgotten who. I thought fxotune set up the built-in FIR filter in the DAA and nothing more. I'm really uncertain how a little filter is going to help with impedance matching, as it's still the same frequency ranges that need to get through to be digitized. I have, however, been known to be mistaken on more than one occassion. :-) fxotune currently only does tuning with the AC impedance functions on the Si3050. If there is continued line-related echo problems, there is always the option of adding the onboard digital hybrid tuning to the mix, but it is unable to fix the problem as much as doing the AC impedance tuning. The onboard hybrid is more for that last step of tweaking, only to be used after doing the line adjustment for the AC impedance. -- Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] delay before dial on TDM04B
Jerry Geis wrote: Samy, Thanks for the suggestion - however I am confused on the wiki the 'w' stands for: *w*: Allow the /called/ user to start recording after pressing *1 or what defined in features.conf (Asterisk > v1.0.x) This is not a delay of any kind. Jerry [Asterisk-Users] delay before dial on TDM04B *Samy Kamkar* samy at fonality.com <mailto:asterisk-users%40lists.digium.com?Subject=%5BAsterisk-Users%5D%20delay%20before%20dial%20on%20TDM04B&In-Reply-To=43134CDA.4020408%40pagestation.com> /Mon Aug 29 13:07:05 CDT 2005/ * Previous message: [Asterisk-Users] delay before dial on TDM04B <http://lists.digium.com/pipermail/asterisk-users/2005-August/123436.html> * Next message: [Asterisk-Users] New astGUIclient version released 1.1.6 <http://lists.digium.com/pipermail/asterisk-users/2005-August/123437.html> * *Messages sorted by:* [ date ] <http://lists.digium.com/pipermail/asterisk-users/2005-August/date.html#123438> [ thread ] <http://lists.digium.com/pipermail/asterisk-users/2005-August/thread.html#123438> [ subject ] <http://lists.digium.com/pipermail/asterisk-users/2005-August/subject.html#123438> [ author ] <http://lists.digium.com/pipermail/asterisk-users/2005-August/author.html#123438> Jerry Geis wrote: / I am searching for a way to add a 2 second delay before calling out />/ with Dial(). />/ Sometimes I get the message "you must first dial a 1 to place this call". />/ I presume the phone company is missing the first digit pulsed out />/ sometimes. />/ />/ How do I put a 2 second delay after coming offhook and before dialing />/ the digits? />/ />/ Thanks, />/ />/ jerry />/ />/ ___ />/ --Bandwidth and Colocation sponsored by Easynews.com -- />/ />/ Asterisk-Users mailing list />/ Asterisk-Users at lists.digium.com <http://lists.digium.com/mailman/listinfo/asterisk-users> />/ http://lists.digium.com/mailman/listinfo/asterisk-users />/ To UNSUBSCRIBE or update options visit: />/ http://lists.digium.com/mailman/listinfo/asterisk-users / You can prepend a 'w' for a half-second wait which will resolve this problem. e.g., Zap/1/w19007529269 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Jerry, Check out: http://www.voip-info.org/tiki-index.php?page=Asterisk+Zap+channels Note this line on the page: /phonenumber/, if present, specifies which telephone number you wish to be connected with. Note that this makes sense only when you are dialing a telephone line (an FXO or PRI interface), not an internal extension. Within the phone number, you may use the special modifier *w* to indicate a half-second pause. You might want to use this to wait for a dialtone or for a pause while dialing digits. You may also use the special modifier *c* to allow for clear channel connections between PRI ports. w = half-a-second wait So, Zap/1/13105551212 would be a 2 second wait. However, I've dealt with a lot of phone providers and none have ever required more than a single half-a-second wait for them to begin detecting the DTMF tones, so you should be good with one 'w'. -samy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] delay before dial on TDM04B
Jerry Geis wrote: I am searching for a way to add a 2 second delay before calling out with Dial(). Sometimes I get the message "you must first dial a 1 to place this call". I presume the phone company is missing the first digit pulsed out sometimes. How do I put a 2 second delay after coming offhook and before dialing the digits? Thanks, jerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can prepend a 'w' for a half-second wait which will resolve this problem. e.g., Zap/1/w19007529269 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP providers
I signed up with fonality and so far they've been awesome compared to vonage and voiceglo (the quality is SWEET, maybe I'm just used to vonage) I called their 877 number too and actually got to talk to a knowledgeable person...I asked them for a feature and they added it the same day. BTW, their FAQ page works for me. -Samy >>Roy wrote: >>>Check out Fonality (http://www.fonality.com) >> >>Umm, I did: >> >>Their 877 number yields a "We're sorry, that number has been disconnected." >> >>Their FAQ has a big series of questions, but the links for each question >>are simply a link to the FAQ page itself. >> >>Is this a simulated ITSP? >> >>B. > >Hi, > >I've had some very prompt emails from fonality. They plan on supporting >Asterisk after June 1, 2004. However, they are geared toward residential >service, have no "business class" calling plans, and their residential plans >explicitly forbid business uses. > >Too bad. > >Michael Swan >Neon Software, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
