similar to: Dial timeouts and SIP 302 redirects

Hello List Next question where google did not spit out an unsable answer. When redirecting a call with Transfer, I would like to correctly indicate the reason. I did try this: exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)}) exten => XX,n,Dial(SIP/ZZ) exten => XX,n,set(REDIRECTING(reason)=cfb) exten => XX,n,Transfer(SIP/YY) I did try with 'reason'

Hi Richard Thank you > You need to set more redirecting information [1]. > > In sip.conf send_diversion=yes needs to be in effect. You also need > to setup > the from party id information (at least the from number) to indicate > where you > are redirecting from. You should also increment the redirecting > count. > > Richard > > [1] >

What is CFU and CFNR? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michel Koenen > Sent: Tuesday, August 30, 2005 1:46 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] How to use * and # as part of > numberindialcommand > > > From: "Damon

I have set up a new SwissVoice phone and it can receive calls but I cannot make calls out from it. The setup is simple for now, 2 phones: SwissVoice is ext 7726 and Cisco 7960 (SIP) is ext 7999. I can call from the Cisco phone and it rings on the SwissVoice phone but when I dial from the SwissVoice phone I get a busy tone upon dialing the second digit. The log reads as follows: -- Endpoint

I am Jerry Richmond CEO of ByVolution LLC. We have purched some did's from you that we use to test with, weare going to order our first batch of 250 this week. John Blackman is not with us any more. I need for some one to call me on my cell phone because our office no. 9198270713 to 9198270720 can not except outside calls. Brian Sponaugle tells me he is not getting the help he needs from

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

Hi, Using 1.4.13 is it possible to ignore 302 redirects from sip devices belonging to a queue? For a queue that rings the whole office it doesn't seem very useful to obey a redirect programmed on a phone. It seems this was the default behaviour in 1.2. Thanks,

Has anyone used the Swissvoice IP 10S (www.swissvoice.net) VoIP Phone with *? Adrian -- Adrian Walker adrian@digitaltraffic.co.uk ======================================================================= This email has been scanned for Virus infection by MessageLabs For more information please contact messagelabs@atomwide.com

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make it work properly....i can't activate the forward in the phone, is quite confuse:

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

I believe I''ve found a bug in which a custom HTTP code handler in the code generated by form_remote_tag does not get called under IE6. The below example simply redirects the browser to the value in the location header when it gets back a 302 redirect: #### start code #### <%= form_remote_tag :url => {:action => "comment", :id => @article},

Hi, Trying to use Ajax.Updater to update a screen element. Works well, except in one place where I want to use a 302 Redirect page. This works absolutely fine in the Browser, and takes me to the page I want to see. However, when calling using Ajax Updater, I get a message like this: This document you requested has moved temporarily. It''s now at http://.... The on302 event fires,

Does MGCP transverse NAT? Seeing as the only decent yet cheap IP phone is the Swissvoice, it would be rather helpful.

I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line => aaln/1 The portion of extensions.conf is: exten => 3001,1,Dial(MGCP/aaln1,20) exten => 3001,103,Hangup

hallo, does anybody successfully managed to get swissvoice ip10s with h323 firmware work with asterisk ? mgcp firmware works fine, but with h323 i'm still getting one way audio. regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

Dear Friends, We have implemented "Asterisk" in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost? Thank you. Regards, Chandra. --------------------------------- Ring'em or ping'em. Make PC-to-phone

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia

Hi chan_capi users, this thing is awesome, no delays like in modem_i4l! Plus, it got those nice ISDN features. Here's my question: Does my service provider (Deutsche Telekom) have to provide me with these Services (CD, ECT)? (the Readme in 0.2.5 says "does not relay on service CD") I know, that I don't have CFU,CFNR,CFBS (which I would have to order seperately). How likely

I am trying to give users the option to query our accts. payable database by supplying their PO number. I able to write queries via perl->DBI->mysql but have no idea how to get * to do it from the IVR. Is this possible? Can anyone point me in the right direction for help or examples? Thanks, Richard --------------------------------- What are the most popular cars?

Hi Does anyone have a howto on how to set one of these up on Asterisk or Trix box please? I can make it SIP or MGCP so whatever you have ;-) I have found one page but it isn't really a howto setup Thanks in advance Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: