similar to: How to route SIP provider without DID

Hi! I am getting inbound caller ID fine bout not out. I am in Sweden and suing Rixtelcom /POrt80 as provider. anyone knowing what is wrong? Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices. Regards Mattias -- Mattias Andersson -------------------------------- Storskiftesv?gen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38

Hi, Is it possible to "regret" blind transfer while its ringing (not answered)? Thank you! Best regards HB

Hi, Is there an affordable HW solution to do a door phone on *? I do not mind using the solder iron to modify an existing door box. Thank you! Best regards HB Norway

Hi, My Asterisk systems runs like a dream with mISDN, SIP and even and old Digium board. But have almost in every conversation some irritating DTMF being generated. The seems to be just as often from all trunks but are worse if noise load speaker in other end. Any good advices? Where to look for forgotten DTMF detection settings? Thank you! HB

Thank you, very interesting! As I understand the Digium card is used as a interrupt source for Asterisk? Is there a diagnostic tool available ? Anybody else experienced a simmialr problem? Thank you! HB > From: > covici at ccs.covici.com > Date: > Sat, 12 Dec 2009 19:04:23 -0500 > To: > Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

Hello, I've been looking at the DialPlans by some poeple using Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to create Outbound routes etc, does using the web admin give the same effects? When I add a SIP Trunk with my Sipgate settings and use a pattern of "8|." to place all calls with a 8 prefix tot he sipgate account the softphones dial the number, the Asterisk

Hello list, I've seen the recent discussions documenting problems with lm. I have encountered the following problem. I use WinXP Pro with service pack 1, and R 1.9.0, on a XEON 2GHz, with 1GB of RAM. > eff.fro std.dev mean NSTRDSP 7.403749e-01 1.215686e-01 CPFGEP 9.056763e+00 1.815686e+00 WSWOLF 4.703588e+05 1.112832e+05 NPILGRIM 1.017640e+06 2.134335e+05

Hi, I would like some advice from you on how to configure a multi line phone the best way! So far I have given the phone 4 sip accounts one for each line, this is a lot of work and gets messy. Is it a better way to do this? Thank you! Best regards Helge-Bj?rn

Hi Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account. www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations. Many Thanks Scott

Hi Any solution for connecting Asterisk to Skype without using fsx/fxo hardware ? Best Regards HB Norway

Hi! I tested two different implementations of the robust MCD estimator: cov.mcd from the MASS package and covMcd from the rrcov package. Tests were done on the hbk dataset included in the rrcov package. Unfortunately I get quite differing results -- so the question is whether this differences are justified or an error on my side or a bug? Here is, what I did: > require(MASS) >

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==

I am looking for a voip provider that provides good rates (below 5 cents/min or unlimited) to UK NCFA numbers. Braodvoice advertises they do unlimited to NCFA but does not have the ability to actually termiate those calls as per the CTO Nathan Stratton, and last he said they dont even have contracts in place to get service provisioned for that. As such I am looking for another provider to take

Hi all, I'm new to the forum. Oh no....newbie question coming, I hear you all cry! I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've

Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. The critical thing is that DTMF must be correctly passed 100% of the time, unlike Sipgate, my current (free) provider, whose DTMF detection/passing is not at all reliable, making it useless for a virtual receptionist scenario. I

Friends, i) I am new to R. Kindly suggest some resources that has examples of plotting with R. ii) How to set number of tick marks and labels, i have a x axis ranging fro 1 to 21. By default R shows the tick marks at 5, 10,15,20. How can i change this. Thanks, Bala [[alternative HTML version deleted]]

Hi I have installed, asterisk , with postgresql. it 's the view of extensions table: didex=# select * from extensions order by id desc limit 5; id | context | exten | priority | app | appdata | description

Hi I have installed the http://asteriskathome.sourceforge.net/ with a Digium card with no problems, very good ! Now I want to install my Billion PCI ISDN card (HFC based) in TE mode. I get a little confused with Isdn4Linux, ZapHFC HIAX and the need to install Capi ! Please suggest best and easiest approach ? Thank you ! HB Norway

Hi all, We currently have 2 Asterisk boxes and we pass calls to a fro. All works great except we now need to pass variables between them. For example now on box 1 we have: exten => _23XX,1,SetVar(Foo=1234) exten => _23XX,2,Dial(SIP/${EXTEN:0}@Box2) When the call dials into Box 2 the variable Foo does not get passed... Does anyone have any clever ideas? -------------- next part