RE: [Asterisk-Users] Has Nufone gone belly-up

Hi Everybody, In regards to what I see here, this looks like a whole .com flash back. I started a phone company that went belly up (CentreCom, the first Unified Communications company) because of customer service issues, lack of on-line information, and a lack of caring for the customer.

Re: [Asterisk-Users] Incoming SIP matching

I ran some tests and reviewed the source code. It appears that for incoming INVITE messages, Asterisk first checks for [name] entries that match the user portion of the SIP URI in the From: header of the INVITE message.. i.e. if you are calling From sip:[EMAIL PROTECTED] it looks for [123] in the

Re: [Asterisk-Users] Has Nufone gone belly-up

Others on this list have shared their (positive) experiences with your company on this list. In fact the reason I attempted to contact you was because of this list. Are you saying that the Asterisk community should only hear one side of the nufone story? I differ. If there are good things to

[Asterisk-Users] GSM phone to *?

Hi! I want to connect my old mobile phone (Nokia 5110) to * through soundcard and COM port, and use it as personal gate to GSM network. Is there a schema how to connect mic and tel from mobile's handsfree connector to line in and line out of soundcard? Can * do external program (gnokii) to

Re: [Asterisk-Users] Has Nufone gone belly-up

Hi John, You are doing NuFone good by giving such indepth guidence on how to operate a business. Like the Chinese saying, you are putting money into NuFone's pocket...they really should thank you for you making a point here on your complaints. For most of people, they just drop it and move on.

[Asterisk-Users] GSM modems

Hi all, I am interested in interfacing a GSM modem to *. I've seen a few comments about doing this, but I'm not clear whether people have actually made it work. I've used GSM modems for various data jobs, mostly high volume SMS (no, not nasty marketing stuff - high volume solicited SMS :-) )

Re: [Asterisk-Users] rc.local dont works

Ok! Thanks miklos - Original Message - From: Karsten Wemheuer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 24, 2004 9:42 AM Subject: Re: [Asterisk-Users] rc.local dont works Hi Miklos, listas iPfone wrote: Hi ! thanks for the answer.. I use rh9... Sorry,

[Asterisk-Users] Need Europian vendor for Digium hardware.

Must accepts wire transfers and ships to Sofia. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

RE: [Asterisk-Users] Need Europian vendor for Digium hardware.

http://www.digium.com/index.php?menu=resellers#Europe -Original Message- From: Anton Tinchev [mailto:[EMAIL PROTECTED] Sent: 26 January 2004 11:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Need Europian vendor for Digium hardware. Must accepts wire transfers and ships to Sofia.

RE: [Asterisk-Users] Asterisk Indications

Hi! You'll find the Australian tone specs here http://www.acif.org.au/ACIF/files/S002_2001.pdf The document below claims 425*25 for Australian dialtone. http://www.teltone.com/prodmanuals/TLE%20Telephone%20Line%20Emulator,%20Rev%20L.pdf Cheers, Philipp

[Asterisk-Users] He really doesn't care

I'm new to this all, and had never heard of NuFone until someone raised the question of whether they were in trouble. This was a net positive for NuFone because it made me a aware of their existence. The next question in my mind was about their service, which I had to evaluate based on second

RE: [Asterisk-Users] GSM modems

Hi, -Original Message- I am interested in interfacing a GSM modem to *. I've seen a few comments about doing this, but I'm not clear whether people have actually made it work. I've used GSM modems for various data jobs, mostly high volume SMS (no, not nasty marketing stuff - high

Re: [Asterisk-Users] GSM modems

26 2004 13:38 Florian Overkamp : No idea about that, but there are devices that connect a GSM to an FXO/FXS port (I have a Siemens Homestation connected to my X100P at home ;-) or even PRI... I do not want to spend $1k, I just want to make home gate ;) So only question is a simple

Re: [Asterisk-Users] Has Nufone gone belly-up

On Mon, 26 Jan 2004, Jeremy McNamara wrote: Our network and services speak for themselves. If they don't like my attitude after they publicly flame us they can find another provider, I really don't care. And I don't care about your network, your services, or your contributions to

[Asterisk-Users] ZAP Problems

Hi all, Since my upgrade to CVS dated 14-01-2004 I am unable to call or receive calls through my ZAP channel. When calling out I get the following message: - WARNING [155667]:app_dial.c:527 dial_exec: Unable to create channel of type ZAP In zaptel.conf fxsks=1 loadzone=uk defaultzone=uk In

RE: [Asterisk-Users] Asterisk Indications

On Mon, 26 Jan 2004, Philipp von Klitzing wrote: The document below claims 425*25 for Australian dialtone. http://www.teltone.com/prodmanuals/TLE%20Telephone%20Line%20Emulator,%20Rev%20L.pdf So does the ACIF document quoted earlier (in fact it mentions dial tone can be 425, 425*25,

Re: [Asterisk-Users] OH323 doesnt hear ringing

I have simillar situation. I have Asterisk running with a combination of SIP and H323. With a SIP clients are connected, with H323 my routers are connected. "When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone

RE: [Asterisk-Users] GSM modems

Hi, -Original Message- No idea about that, but there are devices that connect a GSM to an FXO/FXS port (I have a Siemens Homestation connected to my X100P at home ;-) or even PRI... I do not want to spend $1k, I just want to make home gate ;) My kit cost were: X100P:

[Asterisk-Users] Digium FXO Card

Hi, I wish to know if GNUGk can work with * running as a gateway with the Digium FXO card. Kindly share your experiences in case there are some issues which one must know before going in for such a setup. Also, I've been reading about the DialTone detection capability by the hardware in different

Re: [Asterisk-Users] Has Nufone gone belly-up

Don Feuer wrote: Hi Everybody, In regards to what I see here, this looks like a whole .com flash back. I started a phone company that went belly up (CentreCom, the first Unified Communications company) because of customer service issues, lack of on-line information, and a lack of caring for the

[Asterisk-Users] SIP - fax / voicemail

Hi, Just to clear things out.. Can asterisk transmit faxes over IP ? If not, are there any works being done towards implementing t.38 on asteisk ? Also dialing in from a mediatrix fxs sip gateway to voicemail, asterisk does not see the digits entered after mailbox prompt. I have dtmftone

Re: [Asterisk-Users] Has Nufone gone belly-up

And I don't care about your network, your services, or your contributions to Asterisk. Your behaviour in this matter is like that of a toddler in a sandpit, throwing sand back at the other kids then screaming they started it. Grow up. Your prospective customers have. echo nufone.net killfile

[Asterisk-Users] app_queue and dialplan

Hello, I`m trying to achive this: 1. when the initial call comes in it is served by a small queue with short timeout so that at first caller hears only ringing 2. if nobody answers the call at that time or the queue is all full the call goes to the Playback the message ( please hold bla bla

[Asterisk-Users] Know if a call is answered

Hello: I have an asterisk server answering SIP calls. Whenever a call comes, asterisk answers, plays a gsm file (information) and dials to another SIP phone. Using asterisk Master.csv file I only have one record, and don't know if the second call is answered. I only know this if: - The called

Re: [Asterisk-Users] rc.local dont works

Hi Miklos, I have the same problem here in RH90 - have you found any solution? Or does anybody else know why (safe_)asterisk does not start using rc.local? (normally I start * as root user) Cheers Jeroen ___ Asterisk-Users mailing list [EMAIL

Re[2]: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller

Hello Rich, Sunday, January 25, 2004, 8:01:25 PM, you wrote: RA It would probably help if you used a packet sniffer (eg, ethereal) to look RA at the traffic, or at least provide the list with a useful clue other then RA it doesn't work. RA same here, when i recive an

Re: [Asterisk-Users] OH323 doesnt hear ringing

Aaron Martin wrote: I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing.

RE: [Asterisk-Users] Asterisk Indications

I am really having trouble with this. I have been making changes to indications.conf, but my changes are not taking effect. I have shut down Asterisk, re-run ztcfg, reloaded the zaptel modules, all to no avail -- I get the same tones consistently... What should I be doing to bring in a

Re[2]: [Asterisk-Users] Has Nufone gone belly-up

Hello John, Sunday, January 25, 2004, 11:36:55 PM, you wrote: JB I tried a couple times to talk to them about service. How much it costs, JB how it works, etc. Just common stuff you might find on a website. I left a JB message and nobody returned my call; I went with voicepulse instead. JB

[Asterisk-Users] Questions regarding new echo cancellation features...

I notice the zaptel Makefile option the mark2 option KFLAGS+=-DAGGRESSIVE_SUPPRESSOR is now gone. Does simply adding these options still compile in a certain echo can - or is there an other method of activating a particular can. I have not had to update my machine that is connected to pstn for a

[Asterisk-Users] app_queue and dialplan

Hello, I`m trying to achive this: 1. when the initial call comes in it is served by a small queue with short timeout so that at first caller hears only ringing 2. if nobody answers the call at that time or the queue is all full the call goes to the Playback the message ( please hold bla bla

[Asterisk-Users] X-Lite Asterisk: Speex iLBC not working?

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When

Re: [Asterisk-Users] MGCP Problem.

Hi. Thanks for the tip. Now i'm able to recognize the endpoint MGCP. But i can not place a call. I'm attaching the debug from the asterisk and two configurations files (mgcp.conf and extension.conf). For my first test i just want to call the voice mail, that has the 112 extension. The

Re: [Asterisk-Users] Has Nufone gone belly-up

On Mon, Jan 26, 2004 at 12:40:57AM -0500, Jeremy McNamara said: David Liu wrote: I don't use Nufone. But just seeing Jeremy's reply make me want to say something. As an outsider, if the attitude is No messages were ever received from you, thus we never called you back. or How quickly you

[Asterisk-Users] R2 support

Hi All, We have successfully finished implementing R2 support for *. Drop me an email off-list if you want to test it. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To

RE: [Asterisk-Users] Incoming DID call Voice Problems

I have an updated question on this one. It seems that only inbound long distance calls (calls from outside the local calling area) on our DID trunk have one-way voice. I have my adtran 750 fxs lines configured as FXS Loopstart with all the defaults. Again, the problem is that once the call

Re: [Asterisk-Users] Incoming SIP matching

Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to have dtmfmode=rfc2833. However, incoming FWD calls from the dialup access numbers (such as libretel) need to have dtmfmode=inband. To solve this problem, I created a second FWD account and configured sip.conf as follows,

[Asterisk-Users] Re: Has Nufone gone belly-up

Jeremy McNamara [EMAIL PROTECTED] said: Who says we don't run a trouble ticketing system? We just don't spam you with the lame auto-response message. One guy's spam is another guy's confirmation that his message has landed and that he now has a reference (ticket#) for follow-up communication.