search for: psvasterisk

Hi guys, How's things going ? Got a bit of a weird one here that I've been unable to solve. I have a Panasonic PBX linked to a Sirrix Quad BRI card that is running in TE (ptp) mode in a Asterisk box - this then links through Internet to another Asterisk box via IAX2. When a user on the Panasonic PBX system dials the extension of my Sirrix Asterisk box, Asterisk answers and says

Hello, more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart.... Currently I will have to answer the line to do that. Is there a way to do this with asterisk? Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@luxadmin.org Internet Service

Hello All, I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using Asterisk@Home 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. >From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Does anyone know how to resolve this problem? Thanks, Wiley

Is there a soft phone for PocketPC or iPaq? If not, is someone working on it? I will be more than willing to contribute my mite if needed. Thanks, -- sudhir

...Sam Lex Lethol <lethol@gmail.com> wrote on 23/09/2004 15:31:39: > I tried the xten one and didn;t like at all.. > > Havent tried to SJPhone, but my guess is that it has better support. > > Lethol > > On Thu, 23 Sep 2004 08:13:10 +0200 (CEST), Peter Svensson > <psvasterisk@psv.nu> wrote: > > On 22 Sep 2004, Sudhir Kumar wrote: > > > > > Is there a soft phone for PocketPC or iPaq? If not, is someone working > > > on it? I will be more than willing to contribute my mite if needed. > > > > Xten has a product, possibly stil...

-----Original Message----- From: Peter Svensson [mailto:psvasterisk@psv.nu] Sent: 25 November 2004 10:54 >> They work just fine if your pstn provider is at all serious. If not, >> switch. They don't belong in the pstn business anyway. BT so you probably have a point. To be fair to BT I'm not saying that the ISDN does drop during power outag...

I am have same issue with PRI and overlap dialling is not enabled. Stuart -----Original Message----- From: "Peter Svensson"<psvasterisk@psv.nu> Sent: 01/02/05 16:55:52 To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] Zap channel occasionally misses dialing thefirst digit On Tue, 1 Feb 2005, Andrew Kohlsmith wrote:...

Hi, I wonder which PRI interface card is most stable and supported for EuroISDN and Asterisk ? Are they stable enough ? Any tips ? Thanks in advance, regards, Rob.

Hi All, I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the pricing for these in Europe, so I'd like to hear from people here whether that is a reasonable price for them? TIA & BRgds -- Francesco Peeters ---- GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit

...terisk is the CPE how would I set it to be be the opposite of the serving central office? PRI b-channel sequence ordering in Asterisk can't be as simple as changing the zap group prefix in the Dial() command can it? Ie. g vs. G? > -----Original Message----- > From: Peter Svensson [mailto:psvasterisk@psv.nu] > Sent: September 29, 2004 12:23 AM > To: dboyd@fullmoonsoft.com; Asterisk Users Mailing List - > Non-Commercial > Discussion > Subject: RE: [Asterisk-Users] PRI D-channel signalling error? "Ring > requested onchannel 0/1 a lready in use on span 1. Hanging up owner.&...

Hi, I try to connect my asterisk box with a classic pbx (Siemens). I have a T100P E1 card. modprobe zaptel and wct1xxp ok ztcfg -vv ok I can run asterisk and show the channels but the led still flash red My zaptel.conf : span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 dchan=16 My zapata.conf : ;; E1 switchtype = euroisdn signalling = pri_net group = 1 context=e1 channel => 1-15,17-31

<FONT face="Default Sans Serif, Verdana, Arial, Helvetica, sans-serif" size=2><DIV>I have been playing with getting the sample.call file to work by dropping it into /var/spool/asterisk/outgoing.&nbsp; The process works to the point of calling the desired number and plays the message.&nbsp; The problem is that the message starts playing almost immediately, so if the

I'm all up for reading and looking round for people in the same boat to try and solve the issue together, but there appears to not be large community yet, just the asterisk mail lists. I got Asterisk working with X-Lite great now for internal calls and also calling land line numbers etc. The two problems i'm currently having are: 1. When someone calls in the phones ring 3 times then

Hello, I am trying to make a call from our PABX to Asterisk on PRI interface. How can i configure Asterisk to enter the overlap receiving state if the complete number is not obtained in setup message. Looking forward to any help in this regard Regards Nauman Bin Ali __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection

Hi! I'm trying to setup a Wildcard TE110P with a PRI in The Netherlands. I get a Red Alarm on the line. Is there any way of debugging this? I've tried some configs that should work but without success. Is there any way of telling if the cabling is correct or what else the problem could be? Thanks!! Remco

I have read most of Eicons information on Q.SIG, and I am able to load the Q.SIG protocol (instead of ETSI for example). No strange logging in divacrtl mlog. But how do I tell Asterisk to understand Q.SIG? My PBX is configured for QSIG, but I cannot see anything on my trace when trying to make a call via the S0(Q.SIG) Janne > -----Ursprungligt meddelande----- > Fr?n:

We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution

I am using a HFC-S card in nt mode with zaphfc driver to connect an internal isdn bus. I would like to signal an incoming call on, let's say, 4 phones. Right now I use: Dial(Zap/g1/21&Zap/g1/22&Zap/g1/24&Zap/g1/23&Zap/g1/29,,t) where g1 are my two isdn channels provided by HFC-S card an the 21,22,etc my internal numbers. When the command is executed however, only the

*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way.

Here is the setup: POTS -> PRI -> Asterisk -> ATA (Fax) The ATA is set to only 711. Asterisk's sip.conf sets this device to only 711. Yet, faxing works less than 50% of the time. I cannot possibly be the only person with this problem. I can't even get app_rxfax to work. It answers but rxfax drops the call 5 seconds into it. If anyone out there has near 100% success with