[Asterisk-Users] ADPCM support with RECORD FILE
I want to record audio in ADPCM format. According to the show codecs output of Asterisk, it looks like it supports adpcm. But I do not know what to tell the RECORD FILE directive in my AGI script. The RECORD FILE command usually has this form: RECORD FILE filename format timeout [BEEP] It records fine in WAV or GSM if I enter wav or gsm for the format, but when I try adpcm, it gives me the error: No such format 'adpcm' Does Asterisk support RECORD FILE into adpcm(I need an audio format of better quality than GSM, but with smaller files than typical WAV). Thanks for any help. Gary Franczyk ___ 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
Re: [Asterisk-Users] Need Europian vendor for Digium hardware.
Hi, I'd like to recommend http://www.telappliant.com They responded far faster than a few others I contacted. I placed my order (2 digium E1 cards , 2 ip phones) friday at 15:00. They emailed the invoice 15 minutes later. I paid at my bank at 16:00, and faxed them the proof. The goods arrived monday at 12.00, and they'd sent the right stuff. And all that was international, they are in UK, I'm in Belgium. Excellent service. Richard Bennett. 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: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P on Redhat 9
The TE410P should run fine under both Redhat 9 and Fedora Core 1. My first question is: Are you running a new CVS version? Maybe there have been bugs introduced with all of the recent changes. I'm running under December versions - works ok, except for problems experienced under very heavy call loads, that have been discussed on here many times (still not resolved) But it also sounds possible that you're having basic PRI problems of some sort, perhaps clock sourcing or something similar. Please share your zaptel.conf and zapata.conf, including which span is connected to what. Regards Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gomillion Sent: Monday, January 26, 2004 7:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] TE410P on Redhat 9 I am the proud new owner of a TE410P, and installed it on a RedHat 9 box. After compiling just fine, running like a champ in tests, and having my extensions.conf configured to taste, I went ahead and did a live beta test this past weekend. The phone system stopped responding 3 times. The first time, I got into the asterisk console, vi -rv. I didn't see anything too out of the ordinary, except for lots of messages that all of the lines were busy (even though no calls were up and I have 2 PRI's). I thought it odd, but since I had been tweaking a config file or two, I didn't think too much about it. I just stopped * and Zaptel, and then restarted them, in reverse order. And everything was beautiful. THen, on Saturday night, around 7:00 PM, I tried to call in again (I was calling in every half hour to make sure it was working), it had died. I was in a bad mood, and just restarted the same as I had the day before to get it over with. Finally, on Sunday morning, I tested the phone, and at about 8:00 am * was not responding. I logged in, again with 5 v's, and saw an error message to the effect Span 2 is up. Span 2 is a PRI from * to my Norstar MICS, using a cross-over cable. I think the LBO and signalling are correct, because calls went through it just fine until the 3 times mentioned above. I started by doing a reload, which changed nothing, then a restart when convenient, which changed something... Now, Span 1, 2, 3, and 4 all gave the message that they were up, about 1x per 2 seconds, each. Every 5 or 6 messages I would see that one of the spans was down, but most of the messages were just about being up. I immediately stopped Asterisk and Zaptel, restarted Zaptel, then Asterisk, and everything ran fine through this morning, when I took Asterisk out of the loop (planned end of the beta test). Does anyone have any ideas what might be causing this, and how to diagnose without taking our main phone switch down? I have access to a PRI, but it has no phone numbers that ring into it, so I can connect to real network equipment and make calls, but cannot receive any. Thanks, David Gomillion ___ 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 ___ 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
Re: [Asterisk-Users] Re: Grandstream 100 sidetone
Stephen I think hit it on the mark. I could not figure out how sidetone could be heard as an echo and how it could be loud. On my GS phone the sidetome is very low but with zero delay. The best way to test the side tome is to talk to the voicemail or record application running on a local Asterisk server. Record does not send audio back so anything you e hear is real sidetone. I find it to be decent quality but to low in volume. --- Stephen R. Besch [EMAIL PROTECTED] wrote: dkwok wrote: For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the volume of the voice call. As the sidetone is louder than the conversation it is getting rather distracting. Can the sidetone be calibrated or adjusted? If not, how are people coupling with it? If I'm not mistaken, what you are calling sidetone (the copy of your owm voice that is played back to your earpiece - it's reassuring to hear yourself talk) is actually real echo generated somewhere other than in the phone. It is a network issue, not a phone issue. Read the many, many post on echo and visit the WIKI. ___ 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ 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
Re: [Asterisk-Users] Will Asterisk be supporting RTCP XR in the future?
This article below came up on the newwire. The RTCP XR RFC was published. Will Asterisk be supporting this function in a future release? Does anyone know if any phone vendors are going to be supporting it? Thanks Lee Goodman Our Technology Update this week is about one of those mechanisms. Known as RTP Control Protocol Reporting Extensions (RTCP XR), the technology defines a standard way to detect VoIP call quality by monitoring a variety of key call ingredients such as packet loss, delay and call quality. [snip] http://www.ietf.org/rfc/rfc3611.txt http://www.ietf.org/rfc/rfc3550.txt Sorry for late reply. Yes, if you can get someone to code for this, it would be a pretty cool extension. Just getting normal RTCP in there would be a plus: many hardphones support RTCP (Cisco, Budgetone, Sipura, etc.) and getting some useful data back on each call would be a huge bonus for service providers who need to examine call quality across uncontrolled network segments. Having that stuff in the CDR (or some small subset of it) or in a separate CQDR (call quality detail record) would allow for some very fancy metrics collections. JT ___ 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
Re: [Asterisk-Users] Has Nufone gone belly-up
John Baker wrote: [...] What you (and alot of businesses) don't understand about complaints is that they generally don't come from people who don't want to do business with you, but rather they come from people who do. They point out problems so that they CAN do business with you. That's why you need to listen and respond professionally, not defensively. [...] What you (and a lot of consumers) don't understand is that some customers need to get fired. Some people aren't worth doing business with. Those are the facts. Someone who starts out with a public post of something as ridiculous as Has NuFone gone belly-up because their (even repeated) contact attempts have not gotten a response is likely in that category. You can't make money trying to satisfy everyone. Take me for example: NuFone has my business. They wouldn't if their rates increased due to making pretty user interfaces that I'm not particularly concerned with, or other such things to make the smallest portion of their (prospective) customer base happy. It just doesn't make any business sense to do so. Daryl ___ 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
[Asterisk-Users] RE: Bluetooth discussions (quick glance to some BT products)
There are somes products available claiming to connect a BT headset, a cell phone and a phone land line all together. I've found some : http://www.geekzone.co.nz/content.asp?contentid=2079 http://www.clipcomm.co.kr/ The clipcomm BS-A101 sample price is : $570. It's VoIP land phone that can make/receive a voice call using your CTP cell phone or BT-enabled headset. Maybe the best system is: http://www1.norwoodsystems.com/ It's complete wireless solution that include hot connection to bluetooth hubs when the employee walks in the office. But it's not currently available. Maybe it'll work better with Bluetooth 1.2 wich will be very soon on the market. Here's a reply I got from norwoodsystems : Thanks for your enquiry. I attach a product description to help you understand our solution. We intend to sell the software through resellers and are currently reviewing the business model and packaging. We can achieve a price of under EUR450 per user including software, bluetooth headsets or phone upgrade to CTP enabled GSM phone( Bluetooth Cordless Telephony Profile), VoIP gateway, Bluetooth USB dongles, ISDN Basic Rate Interface, and software installation. ___ 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
RE: [Asterisk-Users] GSM modems
Steve. Flosys makes fixed cellular interfaces. Although our main products come with FXS ports we designed the interface as interchangeable modules. One of our interface modules is a T1/E1 interface (based on an Infineon Falc56). So yes we do support digital interfaces. We also have a TDM interface card which allows to daisy chain several GSM(or TDMA or CDMA) units together to the T1/E1 master unit. P.S: Our FXS module uses the same chipset as the Digium TDM400P card. Regards. Alfred R. Nurnberger _ F L O S Y S Making Communications Flow Tel: +1 (503) 972-9300 Fax: +1 (503) 972-9309 US Toll Free: 1-877-4FLOSYS http://www.flosys.com -- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Monday, January 26, 2004 1:19 AM To: [EMAIL PROTECTED] Subject: [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 :-) ) . These only have analogue ports for voice. Does anyone know of units with digital voice interfaces? Regards, Steve ___ 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 ___ 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
RE: [Asterisk-Users] Asterisk Indications
Chris, thanks very much for this tip: On Mon, 26 Jan 2004, Christopher Lee wrote: My testing involves calling from a SIP handset to a dummy extension setup to answer and playback the tones I want to check. ; Test Australian ringing tones - indications exten = 906,1,Answer exten = 906,2,Wait(1) exten = 906,3,Playtones(ring) exten = 906,4,Wait(12) exten = 906,5,Playtones(busy) exten = 906,6,Wait(5) exten = 906,7,Hangup I set up an extension like this, and found that my indications were being picked up and played correctly by Playtones(). FYI, I found 425*12 to be the closest match to the tones coming from my Telstra line. It sounds like you must have FXS extensions your trying to test the indications on? I don't have an FXS card in my machine to test with, so I'm not sure how it works, but it should still be the same, as a reload definitely re-reads the indications.conf configuration. Yes, I have a handset attached to an FXS port and was using that to see if my indications were changing. They don't. My FXS ports are being handled by the simple switch -- can the tones generated here be altered? Are they supposed to pick up the indications.conf tones? I found 'language=en' in zapata.conf, tried changing that to au but no difference... Cheers, Vic Cross ___ 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
RE: [Asterisk-Users] Incoming DID call Voice Problems
The only thing I can think of in respect to analog DID lines is answer supervision. DID lines provide one way - outbound audio - before answer and cut through bidirectional audio only after answer. But this could happen also outside the local switch so local calls will still have bidirectional audio. Since only the outside caller gets audio, I could imagine that either answer supervision is not given properly or maybe the Wink pulse is not recognized. Something in this vicinity ... See if you can get proper audio if you replace the outside DID line with a regular phone. Check for proper polarity reversal (wink pulse after seizing the line) and constant reversal after answer. Make sure that Ring/Tip is not reversed. DID lines need proper polarity to work correctly. Regards. Alfred R. Nurnberger _ F L O S Y S Making Communications Flow Tel: +1 (503) 972-9300 Fax: +1 (503) 972-9309 US Toll Free: 1-877-4FLOSYS http://www.flosys.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott (7805) Sent: Monday, January 26, 2004 6:32 AM To: [EMAIL PROTECTED] Subject: 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 bridges, the outside caller can hear the person they called, but the inside person can't hear the caller. This happens regardless of the internal technology, SIP, Zap, H323. Could it be possible that inbound long distance calls are signalled different than inbound local calls? Inbound calls on the PRI work flawlessly. Any ideas -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott (7805) Sent: Saturday, January 24, 2004 3:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Incoming DID call Voice Problems Hello All, I am experiencing some intermittent problems with calls coming inbound on my DID trunk. I have 12 DIDs that come into an Adtran 750. From there T-1 to a port on T400P. The problem is that some calls that come in don't seem to bridge properly. Heres what happens. Call comes in on Trunk. Call Routed to correct Zap Channel. Phone Rings. Person Answers phone, but hears nothing but their own echo. Calling party hears everything fine. I have MARK2 enabled in Zaptel driver for echo problems on my PRI line. I can't seem to replicate the problem calling out PRI to the DIDs, or from a cell phone. I can reliably replicate the problem with an offsite customer that calls in. Any idea what may be causing this? Thanks in advance. ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] Questions regarding new echo cancellation features...
I'll answer my own post since I plunged ahead while my original post was stuck in my outbox over the weekend... I notice the zaptel Makefile option the mark2 option KFLAGS+=-DAGGRESSIVE_SUPPRESSOR is now gone. Ok, I found zconfig.h for KFLAGS+=-DAGGRESSIVE_SUPPRESSOR How are people liking conversations with the echotraining enabled on both ends of connections like... remote* iax * pstn whatever My observations (so far) ... the echotraining looks pretty good on the remote end. I'm still suspicious about the pstn end, as an outgoing call is likely to be connected to a pbx, cell network or whatever and the canceller will initializes before your call is connected to a person. -- This message was sent using Monroe-Woodbury's WebMail. http://webmail.mw.k12.ny.us/ This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus. ___ 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
[Asterisk-Users] Not sure what I'm looking for to ask correctly.
I have an * box at my home office. I have my GS 101 at work that registers fine with it. However, when I try to dial voicemail from the phone @ work, it sometimes doubles the digits I enter. Example: I designated 8500 as my voicemail, so when I dial it...and it prompts for mailbox/password and I enter 5000 password 1234. It will double the digits. I see it when watching the CLI, mailbox:55 password:11223344. The speed is around 115ms from the phone to the * box. What do I need to do to prevent this? Sorry if I don't know the exactly terminology. Regards, Joe ___ 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
Re: [Asterisk-Users] rc.local dont works
Hi Jeroen1 I think that´s maybe a bug I really don´t found the problem in my logs, i´m starting it by hand :-( I update you if i can figure it out. regards Miklos - Original Message - From: Jeroen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 26, 2004 11:23 AM Subject: 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 PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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
Re: [Asterisk-Users] AVM C4 with Asterisk in front of an ISDN PBX
Hi! anyone out there running an ISDN pbx behind an Asterisk server with an AVM active card? At this moment Asterisk is connected to the internal ISDN bus of the PBX, but I want to control the calls coming in from the outside, before sending them of to the PBX. Can't be of help here - I have a ISDN PBX *parallel* to Asterisk (Auerswald ets 2106i, 50 at ebay). It serves the analog fax machine, and sits idle as an immediate backup if Asterisk is out of service. The ISDN PBX answers slower than Asterisk, so Asterisk always wins... anyway, now that I have grown more comfortable with * I trust it more and wouldn't necessarily set up things that way again, but it still feels good. ;- Cheers, Philipp ___ 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
[Asterisk-Users] Re: Questions regarding new echo cancellation features...
john wrote: 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 while I don't want to jump into echotraining without a way to quickly enable what has been working for me. How do I enable this mode now? options were moved to zconfig.h ___ 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
[Asterisk-Users] Asterisk Appliance
Ok now that we have a Asterisk server running quite well, we want to put it onto a more appropriate device, i.e. not a beige box computer, but perhaps some kind of embedded linux appliance. Has anyone already done this? Any suggestions on some tidy, small, suitable linux systems to use for asterisk? I.e., somthing that looks like this: https://secure.makonetworks.com//images/main/mako_250_shad.gif
Re: [Asterisk-Users] X-Lite Asterisk: Speex iLBC not working?
If this may be of any use: I'm not an expert but I did the test with the FWD soft phone from X-ten and iLBC SPX don't work. Asterisk wasn't between the connections. Just x-lite and fwd (who is an Asterisk Server?) The soft phone makes the connection but I can't hear any sound. Wim - Original Message - From: Fran Boon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 26, 2004 3:01 PM Subject: [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 forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing iLBC, there is some very garbled noise, but nothing intelligible. Sniffing the packets, I can see that X-Lite Asterisk have chosen differing 'Payload type' numbers: X-Lite: a=rtpmap:97 speex/8000 a=rtpmap:98 iLBC/8000 Asterisk: a=rtpmap:97 iLBC/8000 a=rtpmap:110 SPEEX/8000 According to the Speex RFC, this is acceptable: http://speex.org/drafts/draft-herlein-speex-rtp-profile-00.txt Dynamic payload type codes MUST be negotiated 'out-of-band' for the assignment of a dynamic payload type from the range of 96-127. I'm wondering whether the system is at all case sensitive? From the RFC: When conveying information by SDP [4], the encoding name SHALL be speex. NB Ethereal shows payload-type as being 97 when X-Lite reports iLBC 110 when X-Lite reports Speex, so the Asterisk numbers seem to 'win'. Any light shed on this would be great. Whilst GSM is ok, it would be great to leverage the power of Speex :) Thanks a lot, Fran. ___ 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 ___ 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
[Asterisk-Users] # transfer with IAX2
Hi all, I'm having a bit of a problem using the # sign to transfer when using a soft IAX2 client. Has anyone else experienced this problem or know of a possible work around / fix of this problem. The following is a snippet from my extensions.conf file. This is how the file is setup for inbound calls. When I was using IAX1 based client, I had no problem at all. I simply would hit the # and I would get a voice prompt to transfer; however now when I hit the # I get nothing. The caller hears a tone and the CLI says unable to transfer. [incoming-callers] exten = 5000,1,Dial,IAX2/myclient|40|t Thanks all! AJ ___ 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
[Asterisk-Users] Call Queue wait times
Title: Call Queue wait times Hi Everyone, Is there any specific way to get the current wait time for a queue? If not what is the best way to implement this feature? I would really like to be able to intelligently estimate wait time. Thanks, Matt
[Asterisk-Users] H 323 + Netmeeting test drive
Hi to everyone, I am dealing with my primer Asterisk installation and we are trying to set up a H323 server in order to use Asterisk to place calls between NM clients (also Gnomemeeting). I have a basic extensions.conf file: [general] static=yes writeprotect=no [default] exten = user1,1,Wait,1 exten = user1,2,Answer exten = user1,3,ResponseTimeout,4 exten = user1,4,Hangup exten = user2,1,Wait,1 exten = user2,2,Answer exten = user2,3,ResponseTimeout,4 exten = user2,4,Hangup And I have an even more simple h323.conf file: [general] port=1720 bindaddr=192.168.1.1 tos=lowdelay [user1] type=friend host=192.168.1.2 context=default [user2] type=friend host=192.168.1.3 context=default With this configuration, I managed to see some debug when a h.323 debug command is dropped in the CLI command line, but none of the users see the call. User 1 has Gnomemeeting (me), the other one has NetMeeting. To be fair, I don't know exactly how to set up properly both clients to work fine with Asterisk, so maybe I might be a misconfiguration issue. Of course I have compiled the Pwlib and the OpenH323 modules and set them up, I have no errors nor warnings about these modules, I have a few ones regarding to Oss modules and Iax, but they don't bother me at the moment. Is there any other issue I must pay attention to in order to see calling messages in those VoIP clients?? My Linux test machine (where i run both Asterisk and Gnomemeeting client) doen't have its sond card set up properly and i have some error from Gnomeeting when a make a call, but the other peer does not get any call. Hopefully some of you guys, will so patent to give me some light on that problem?? I need to show something clear to my manager and we'll not have anything to make a decission. Thanks in advance Regards Diego Fernandez PS: Some Asterisk IRC users gave me some ideas about not using H323 and go to SIP, but my first goal has to be make the thing wotk with netmeeting clients or similar (SIP) but free, and afterwards make a decission about other options.
Re: [Asterisk-Users] Zapateller
I have a few (more specific) questions about 'Zapateller': 1) How would you test this?? Would I need a predictive dialer machine like the telemarketers use. OK, I could just wait and see if it seems to cut down the unwanted calls but that's not really a test. 2) I don't understand how I would choose to use the answer option or not. under what conditions would answer by better then not using the option? I asume if I used answer the tones would play in caller's ear and be _very_ anoying to them. Why would I want to do that? There must be some reason are why would the option exist? --- Steve Foy [EMAIL PROTECTED] wrote: Hi, I'm just wondering about 'Zapateller'. How exactly does it work!? I might be interested in employing it at work here, but wondering if anyone's using it? echo*CLI show application Zapateller -= Info about application 'Zapateller' =- [Synopsis]: Block telemarketers with SIT [Description]: Zapateller(options): Generates special information tone to block telemarketers from calling you. Returns 0 normally or -1 on hangup. Options is a pipe-delimited list of options. The following options are available: 'answer' causes the line to be answered before playing the tone, 'nocallerid' causes Zapateller to only play the tone if there is no callerid information available. Options should be separated by | characters -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ 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
Re: [Asterisk-Users] Has Nufone gone belly-up
I hope you are prepared to be mightly flamed when you complain about nufone not responding to your emails grin Nick On Mon, Jan 26, 2004 at 09:09:32PM +1000, Vic Cross wrote: 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. 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 -- Vic Cross ___ 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 ___ 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
[Asterisk-Users] differentiate incoming calls on SIP clients
Hello all, I would like to set-up some direct lines so that when a user of mine answers the phone he/she knows to say the correct intro message, so that we can introduce ourselves as different companies. I have played around with caller ID and can modify that using caller ID Name doesnt seem to work unless it is numeric? Do you have any imaginative solutions for this? Thanks Nick
Re: [Asterisk-Users] OH323 doesnt hear ringing
On Mon, 26 Jan 2004, 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. However, when the H323
client rings a SIP client, there is no ringing sound at all, although as soon as the
called party picks up the phone, everything works fine. This is the entry from my
extensions.conf:
exten = _7[5-9]X,1,Dial(SIP/${EXTEN},20,rt)
exten = _7[5-9]X,2,Playback(vm-nobodyavail)
exten = _7[5-9]X,3,Hangup
I assume that because I havr the 'r' in the dial string, the calling party should
hear a ringing noice. Any ideas?
I had the same problem, depending on which kind of channel I was calling.
Adding an explicit ring like this helped:
exten = _X,1,Ringing
exten = _X,2,Dial(SIP/${EXTEN},20,t)
exten = _X,3,Playback(...)
...
Cheers,
Siggi
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Re: [Asterisk-Users] app_queue and dialplan
Try with: http://bugs.digium.com/bug_view_page.php?bug_id=214 Regards, Gus - Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 26, 2004 11:01 AM Subject: [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 bla) 3. Then the call goes to another queue and he holds while the music-on-hold plays a app_queue trys to reach the next free operator 4. after a timeout in second queue there is a Goto to play the message again and then back into the second queue I have it like this: extensions.conf: exten = 10,1,Queue(q1_short,tn) exten = 10,2,Answer exten = 10,3,Playback(please_hold) exten = 10,4,Queue(q1,t) exten = 10,5,Goto(3) queue.conf: [q1] music = test announce = test_anounce timeout = 40 retry = 3 maxlen = 10 strategy = leastrecent member = SIP/111 member = SIP/112 member = SIP/113 member = SIP/114 member = SIP/115 [q1_short] music = test announce = test_anounce timeout = 15 retry = 3 maxlen = 3 strategy = leastrecent member = SIP/111 member = SIP/112 member = SIP/113 member = SIP/114 member = SIP/115 but the broblem is when the q1_short is full, and the call goes to the q1 it only plays the announce message and and no music on hold is played and again the announce message is played. somehow the music on lod doesn start. What am I doing wrong? I run version CVS-12/01/03-14:50:57 Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ 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 ___ 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
Re: [Asterisk-Users] TE410P/Zaptel
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 23 January 2004 15:08, Tais M. Hansen wrote: How can I configure the TE410P card to act as master instead of slave? Ah, the wiring was wrong. Straight when connected to the Telco, crossover when connected to a local PBX. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAFop42TEAILET3McRAsb+AJ9dPAV/mceZW42zle3qtnc5/jQkSwCfYpTm Rvi+hdo0zpEG5Wlu3nOdQOk= =up+L -END PGP SIGNATURE- ___ 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
Re: [Asterisk-Users] Mailing List Lag
On Thu, Jan 22, 2004 at 11:44:52AM +1100, [EMAIL PROTECTED] wrote: Has anyone from digium looked at why there is a 30 min to 3 hour lag on messages on this list? Continuing this... I sent a message (Subject: Zapateller) yesterday at 17:54 GMT and it only came through to my box today at 08:28 GMT. That's over 12 hours. My box has been up and accepting mail all night as it does every night, so it's not a problem there. Has anyone any suggestions or thoughts about how to improve it? -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ 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
Re: [Asterisk-Users] Zapateller
On Mon, 26 Jan 2004, Steve Foy waxed: I'm just wondering about 'Zapateller'. How exactly does it work!? I might be interested in employing it at work here, but wondering if anyone's using it? I think you can just put it in your dial plan: exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Zapateller exten = s,4,Dial(whatever) It fakes a disconnected number by playing the same tones you would get when dialing, for example, 555-1234. If a computer was calling you, sometimes they are programmed to listen for those tones. No sense keeping the connection with a disconnected number. I mean, if the National Weather Service is trying to call everyone in the path of a hurricane as fast as possible with information about nearby storm shelters and emergency plans, they want their computer to connect to as many valid phone numbers as it can and play their warning message. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ 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
RE: [Asterisk-Users] rc.local dont works
I'm not sure if you're trying to accomplish something specifically by using rc.local, but I use RH9, and I used make config on both asterisk and zaptel and that created the correct init files for me. Starts up perfect every time! Sean -Original Message- From: listas iPfone [mailto:[EMAIL PROTECTED] Sent: Monday, January 26, 2004 1:52 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] rc.local dont works Hi Jeroen1 I think that´s maybe a bug I really don´t found the problem in my logs, i´m starting it by hand :-( I update you if i can figure it out. regards Miklos - Original Message - From: Jeroen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 26, 2004 11:23 AM Subject: 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 PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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 ___ 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
[Asterisk-Users] test list
Hello Asterisk, This is test to the list.. -- Best regards, Frankie ([EMAIL PROTECTED]) mailto:[EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Has Nufone gone belly-up
I'm another outsider just reading this entire thread and Jeremy's replies. NuFone could have the best service on the planet, but I would NEVER do business with a company that has this kind of attitude. People have a choice in which companies they do business with. Why would they do business with a company that get's hostile when problems arise? Jeremy has every right to get hostile in this matter. Here we have someone that claims that the service is bad then everyone gangs up on him and beats the shit out of him on the list. I'm sorry but if you have a problem with NuFone you need to contact nufone directly. I had to call NuFone today and let me tell you it was 2 rings before Jeremy Answered his phone and I learned it was a local cable cut that was causing the problems in my area. (SBC Victim ack EVIL TELCO) Jeremy: just a little hint of advice. Check the attitude. It makes you and your company look very bad. Maybe you just had a bad day: we all have them. You still probably have a chance to recover with a good professional response to the thread (not my message...) Attitude is the spice of life. Learn to look past that and you will be a happy person. Jeremy I have to say I have been more than pleased with the service you have provided me and the company I work for. I will always recommend your service to anyone and have in the past. Try not to stress over it man. Remember everyone, in the end we're ALL human (well except the scum that works at SCO we really don't know what they are yet, we can hope the mars lander finds the rest of their family) I think email puts up some distance between that fact and our temper. I have to be the first to admit I will rip someone a new ass in an email faster than I would even think about it in a face to face situation. So before you pass judgement on someone for their response take this into account because I would have reacted in the same manner Jeremy did but i'm not anythying like that if you talk to me on the phone or face to face. Keep up the good work Jeremy. Thanks, Brian The Bitch West aka bkw_ ___ 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
[Asterisk-Users] festival patch missing in latest CVS or stable build
Hello I downloaded the stable build of * and was not able to find the festival patch in that build. Also i tried from CVS and the same. Can anyone tell me where i can find the festival patch? Or the patch is no more requried. i can just compile festival and it will work?? Thanks in advance Regards Deepak
Re: [Asterisk-Users] TE410P on Redhat 9
Does anyone have any ideas what might be causing this, and how to diagnose without taking our main phone switch down? I have access to a PRI, but it has no phone numbers that ring into it, so I can connect to real network equipment and make calls, but cannot receive any. it might be something different, but previously ive had lots of problems with redhats new thread model and asterisk. it was causing processes to hog resources and generally made my asterisk servers incredibly unstable and often unusable. the problem was solved by running this before starting asterisk: export LD_ASSUME_KERNEL=2.4.1 i now have this in /etc/init.d/asterisk before asterisk starts - so its always run before asterisk starts. it may not be the cause of your problems, but even so i found it was useful and so i hope this helps duncan ___ 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
RE: [Asterisk-Users] Asterisk Indications
Darren, On Tue, 27 Jan 2004, Darren McIntosh wrote: Do you have country=au set in indications.conf? A reload brings the indications.conf changes in for me. Yes, I do. The cadences of all the tones are correct, which tells me that at least part of it is working. The frequencies of the tones, however, seem to be in ignorance of the values I set. Again, this only applies to the tones generated by the simple switch; if I use Playtones(), they are correct. Do I need to specify country=au in zaptel.conf or zapata.conf as well? Here's what I have: zaptel.conf # zaptel.conf loadzone=au defaultzone=au fxsks=1 fxoks=2 fxoks=3 zapata.conf ; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; language=au ; ; Default context ; context=default ; ; Switchtype: Only used for PRI. ; switchtype=national ; rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.5 txgain=0.5 group=1 callgroup=1 pickupgroup=1 musiconhold=default ; ; ; Each channel consists of the channel number or range. It ; inherits the parameters that were specified above its declaration ; signalling=fxs_ks busydetect=yes callerid=asreceived channel = 1 ; Channels on the FXS card signalling=fxo_ks callerid=Test handset 1 221 context=international channel = 2 callerid=Test handset 2 222 context=local channel = 3 Thanks, Vic Cross ___ 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
[Asterisk-Users] testing
Testing once again. bkw ___ 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
[Asterisk-Users] DTMF tone stops working with IAX (voicepulse)???
Hello all, I am using voicepulse DID's to receive calls via IAX to and asterisk IVR dial plan I have put together. The problem is after 3-5mins the system cant pickup the DTMF tones I am sending... I have tried different telephones... It just repeats menu options over and over I have to call back and then it works again for another few mins... Any ideas... iax.conf? issue? Thanks, J.C.
RE: Re[2]: [Asterisk-Users] Has Nufone gone belly-up
Yep.. I am a voicepulse subscriber and can also report they don't answer their email either. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Frankie Gravato Sent: Monday, January 26, 2004 5:42 AM To: John Baker Subject: Re[2]: [Asterisk-Users] Has Nufone gone belly-up ... John Good Luck with Voicepulse they are 0 for 5 this month so far i've notice 5 outages this month alone and getting any email back from them is next to impossible.. -. ___ 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
[Asterisk-Users] List traffic
All of a sudden my list traffic appears to have dropped to a few messages/day the past few days. I anyone else seeing this as well? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] ...I believe in love, its all we've got. - Elton John ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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
[Asterisk-Users] IAX2 / SIP testing
Hello I am interested in testing the voice quality with another * setup in India. I am planning to setup a * server at India and would like to know whether the audio quality would be good enough to make freequent calls. anyone willing to help please let me know and i can test in your convenient time. Thanks in advance Regards Deepak
[Asterisk-Users] Cisco 7960 Problems
Has anyone ever seen these errors generated by a cisco 7960? none of our other brand phones seem to generate these erros: Jan 27 21:54:07 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)Jan 27 21:54:08 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)Jan 27 21:54:12 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)Jan 27 21:54:14 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unable to determine sequence number from ''Jan 27 21:54:18 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unable to determine sequence number from ''Jan 27 21:54:22 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unable to determine sequence number from '' Thanks! Any feedback would be appreciated :) Chris
[Asterisk-Users] CDR records on call transfer
HI, Once the call enters into asterisk and then that call gets transferred, cdr does not record CLID and SRC field data for the transferred call. Is this a bug or I am missing something? Ta SJ ___ 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
[Asterisk-Users] IAX2 / SIP testing
Hello I am interested in testing the voice quality with another * setup in India. I am planning to setup a * server at India and would like to know whether the audio quality would be good enough to make freequent calls. anyone willing to help please let me know and i can test in your convenient time. Thanks in advance Regards Deepak
[Asterisk-Users] festival patch missing in latest CVS or stable build
Hello I downloaded the stable build of * and was not able to find the festival patch in that build. Also i tried from CVS and the same. Can anyone tell me where i can find the festival patch? Or the patch is no more requried. i can just compile festival and it will work?? Thanks in advance Regards Deepak
[Asterisk-Users] cdr_addon_mysql compile error
Having some trouble building cdr_addon_mysql. I've installed mysql and mysql-dev and all related rpms on a RH9 box. The box is up and running *. I've checked out the asterisk-addons but at make install, I receive the following error: ./mkdep -fPIC -I../asterisk -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:113: structure has no member named `userfield' cdr_addon_mysql.c:114: structure has no member named `userfield' cdr_addon_mysql.c:114: structure has no member named `userfield' make: *** [cdr_addon_mysql.o] Error 1 any ideas? thanks, wr ___ 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
RE: [Asterisk-Users] My experience with one IAX termination provider and one SIP provider
Excellent note. I took a look at Broadvox and filled out their little form. The result was This is an automatically generated Delivery Status Notification. Delivery to the following recipients failed. [EMAIL PROTECTED] Oh well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Graves Sent: Monday, January 26, 2004 7:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] My experience with one IAX termination provider and one SIP provider In the wake of the recent NuFone thread I'd like to offer up a brief summary of my limited experience with both Vonage and VoicePulse Connect. I live in Houston TX and for about a year had a Vonage line as my entre into VOIP. Back then, as now, I'd go to some length to avoid Southwestern Bell. I work for a UK based company and my operational territory is The Americas thus I use a lot of long distance, mostly to the US, Canada the UK. The Vonage service was very reliable and I could usually not tell the difference between one of my POTS line and the ATA connected Vonage line. Their service was in my opinion exemplary. One problem with Vonage for me is the billing model. When I travel (a lot) paying a monthly fixed price makes no sense. Some months I use very few minutes, others 2000+. Also, I couldn't get a real second line without incurring another significant monthly fee. Still, they offered local DIDs and I used them happily for a year. In looking at * I was attracted by the prospect of control, features, moh, aa, etc. So I built an * server and bought a few SNOM phones. This displaced my 4 line/8 ext Panasonic office phone system. Connecting the Vonage provided ATA to an X100p didn't seem to make much sense. Needless D/A and A/D steps. Passing SIP through my router meant opening up multiple ports which I really didn't want to do. So I sought out a provider who offered IAX terminationand ended up with VoicePulse Connect (VPC). I've only been using VPC for about a month. The quality of calls over VPC is not quite as good as I had with Vonage. I suppose that could be in part a codec issue. At present I use GSM to connect to VPC to save some bandwidth. I'm open to suggestions as to how to tweak the connection. The service was easily setup. From the point of account creation to initial call was about 30 minutes, largely due to my having to edit the dialplan. No waiting for hardware to ship. The account was up and ruinning immediately even as I worked on it near midnight. The service has thus far been reliable. It supports multiple (up to 6) simultaneous outbound calls which I have used to conference several co-workers. The management imterface on their web site is not as feature rich as Vonage, but it gets it done for me. I like the pay as go, no monthly fee billing model. It's so convenient that even if I use another provider I will keep VPC as a fallback/alternate in my dialplan. One problem however is their lack if DIDs in my area, so I only use it for outbound calling. POTS lines for inbound. Later this month I will likely drop two of my incomming POTS lines as the * server takes on more of the load. That's a vote of confidence in VPC, but I'm still looking for options. If NuFone provides Houston DIDs then they're likely my next experiment. Broadvox Direct, while interesting, is a non-starter due to their flat rate billing model ala Vonage. I also hope to add a wifi sip phone to fill in the last void (cordless) left by the departure of the Panasonic system...if ever they become available. I am eager to hear about others experience with the various providers, even IP Centrex providers. This is just my recent experience. As usual YMMV. Michael Graves P.S. - what to do with my Vonage-crippled ATA? -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] With us or against us isn't a policy worthy of a democratic superpower. -- Zbigniew Brzezinski, Former US National Security Advisor ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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 ___ 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
Re: [Asterisk-Users] Mailing List Lag
Steve, I have been working most of the day to get this problem solved. Thus far everything should be returning to normal and problems like this should never happen again. Right now it has about 3 hours of posts to get out and some were lost by accident during this mess. So if you see a post of yours not on the list please repost it. (but give it another few hours to get caught up.) Thanks, bkw On Tue, 27 Jan 2004, Steve Foy wrote: On Thu, Jan 22, 2004 at 11:44:52AM +1100, [EMAIL PROTECTED] wrote: Has anyone from digium looked at why there is a 30 min to 3 hour lag on messages on this list? Continuing this... I sent a message (Subject: Zapateller) yesterday at 17:54 GMT and it only came through to my box today at 08:28 GMT. That's over 12 hours. My box has been up and accepting mail all night as it does every night, so it's not a problem there. Has anyone any suggestions or thoughts about how to improve it? -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ 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 ___ 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
Re: [Asterisk-Users] Test
Hi, a Virus was sent from this account to the Asterisk-Users mailing list... scan your computer for virus!!! Ronen - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 26, 2004 7:29 PM Subject: [Asterisk-Users] Test The message contains Unicode characters and has been sent as a binary attachment. ___ 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
Re: [Asterisk-Users] Incoming DID call Voice Problems
And check the firmware revision in your Adtran. I believe current is L36. Alfred R. Nurnberger wrote: The only thing I can think of in respect to analog DID lines is answer supervision. DID lines provide one way - outbound audio - before answer and cut through bidirectional audio only after answer. But this could happen also outside the local switch so local calls will still have bidirectional audio. Since only the outside caller gets audio, I could imagine that either answer supervision is not given properly or maybe the Wink pulse is not recognized. Something in this vicinity ... See if you can get proper audio if you replace the outside DID line with a regular phone. Check for proper polarity reversal (wink pulse after seizing the line) and constant reversal after answer. Make sure that Ring/Tip is not reversed. DID lines need proper polarity to work correctly. Regards. Alfred R. Nurnberger _ F L O S Y S Making Communications Flow Tel: +1 (503) 972-9300 Fax: +1 (503) 972-9309 US Toll Free: 1-877-4FLOSYS http://www.flosys.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott (7805) Sent: Monday, January 26, 2004 6:32 AM To: [EMAIL PROTECTED] Subject: 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 bridges, the outside caller can hear the person they called, but the inside person can't hear the caller. This happens regardless of the internal technology, SIP, Zap, H323. Could it be possible that inbound long distance calls are signalled different than inbound local calls? Inbound calls on the PRI work flawlessly. Any ideas -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott (7805) Sent: Saturday, January 24, 2004 3:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Incoming DID call Voice Problems Hello All, I am experiencing some intermittent problems with calls coming inbound on my DID trunk. I have 12 DIDs that come into an Adtran 750. From there T-1 to a port on T400P. The problem is that some calls that come in don't seem to bridge properly. Heres what happens. Call comes in on Trunk. Call Routed to correct Zap Channel. Phone Rings. Person Answers phone, but hears nothing but their own echo. Calling party hears everything fine. I have MARK2 enabled in Zaptel driver for echo problems on my PRI line. I can't seem to replicate the problem calling out PRI to the DIDs, or from a cell phone. I can reliably replicate the problem with an offsite customer that calls in. Any idea what may be causing this? Thanks in advance. ___ 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 ___ 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 ___ 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 ___ 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
[Asterisk-Users] Distinctive ring Issues
Hello all! We have a PSTN line with four numbers calling into it. There is distinctive ring on these lines. They are are follows: 1. standard ring 2. short ring 3. long ring 4. short ring, long ring, short ring Based on the information I have been able to find, I have created the following entries in my zapata.conf file, to try and weed out some of the timings: dring1=95,0,0 dring1context=dist_ring1 dring2=95,325,95 dring2context=dist_ring2 dring3=325,0 dring3context=dist_ring3 ; If no pattern is matched here is where we go. context=dist_ring0 channel = 1 I am assuming that 95 ms is a short ring and 325 ms is a long ring. In my extensions.conf file, I have the following contexts defined: [dist_ring1] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7002,SIP/ringwald) [dist_ring2] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7003,SIP/ringwald) [dist_ring3] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7005,SIP/ringwald) [default] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7001,SIP/ringwald) No matter which number I dial, I always get the [default] context on answer. Can anyone shed any light on what I am doing wrong? The PSTN line is through Qwest Business, and uses US format distinctive ring tones. Show version in the asterisk console returns: Asterisk CVS-01/27/04-19:07:39 Thank you in advance for any help! Steve Ringwald ___ 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
[Asterisk-Users] ztdummy won't compile
For both the latest CVS and the packaged distribution. I dug up the patch that Jean-Denis Girard posted to the list fall 02 to fix the usb problems with the 2.4.20 kernel but that won't apply (not a big suprise). Is this code already integrated? Any other know issues with ztdummy? This is on Suse 9 (kernel 2.4.21-99 SMP). Here's a quick snip of the errors that are thrown when I attempt to compile it. gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux/include/linux/modversions.h -DSTANDALONE_ZAPATA -c ztdummy.c ... cc ztdummy.o -o ztdummy /usr/lib/gcc-lib/i586-suse-linux/3.3.1/../../../crt1.o(.text+0x18): In function `_start': ../sysdeps/i386/elf/start.S:98: undefined reference to `main' ztdummy.o(.text+0x7): In function `init_module': : undefined reference to `uhci_devices' ztdummy.o(.text+0x38): In function `init_module': : undefined reference to `kmalloc_Rsmp_93d4cfe6' ztdummy.o(.text+0x6f): In function `init_module': : undefined reference to `sprintf_Rsmp_1d26aa98' Thanks, Chris ___ 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
[Asterisk-Users] Junk calls from FWD numbers
My Asterisk server registers two FWD numbers. On average I get about one call a day from someone calling from an FWD number and leaving a pointless, under 10 second message. It's easy to see who these people are if I look in my CDR file I can see thier name and number. They seem to be new FWD users, likely who've just downloaded FWD's Xten softphone and then dial some random FWD user (me) to try it out. I wonder if these same people when they first got a POTS phone installed in thier home got out the white pages and dialed randomly asking anyone who'd answer Hi does this work? can you hear me? Question: Does everyone with an FWD number get these junk calls or am I the only lucky one? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ 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
Re: [Asterisk-Users] Re: retrans_pkt: Maximum retries exceeded on call
Try setting canreinvite=no in sip.conf. It might be that attempts to natively bridge the voice streams are failing. On Saturday 24 January 2004 23:26, Chris Wilson wrote: Hmm, The host seems to be good, I have no firewall rules in place at the moment for the local network, and everything is consistantly reachable. it seems to only happen when a call is hung up/initiated, and when the program is first started...if that might provide any insight. Thanks!:) Chris - Original Message - From: Doug Meredith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 24, 2004 4:02 PM Subject: [Asterisk-Users] Re: retrans_pkt: Maximum retries exceeded on call Chris Wilson [EMAIL PROTECTED] wrote: Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Just guessing here, but it sounds like Asterisk sent a request, didn't get a reply, sent again, didn't get a reply, and so on until it hit an internal limit. If my guess is correct, I suppose there could be many causes, including: * Target host down * No path to the target * Firewall blocking traffic * Target host not running SIP, at least on the targeted port. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ 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 ___ 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 ___ 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
[Asterisk-Users] Re: [Iaxclient-devel] Introducing Firefly
Hi, - Original Message - From: Adam Hart I believe you can connect using a standard asterisk box but you'll miss out on the extended features. It is a standard IAX client or a softphone using a modified/non-standard version of IAX2? I have tried to register with my own * box without success. The network runs on iLBC so unforunately it won't work with most IAX2 clients (unless you get * to translate) There is any specific/technical reason not to use GSM as another possible codec? If you don't wish to connect to the firefly network, click cancel when it asks you. When you have setup multiple servers and try to dial a number which of the servers is used to place that call? I have defined just one (my box) and then tried to call an existing extension, but it doesn't work. Best regards, Dan ___ 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
[Asterisk-Users] testing
testing ___ 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
Re: [Asterisk-Users] Has Nufone gone belly-up
[EMAIL PROTECTED] wrote on 01/26/2004 01:12:09 AM: You're right, Jeremy. I made up the whole thing. I went out of my way to concoct a story about how I wanted to do business with you, but was unable to figure out how on your website, so I called and left a message and didn't get a return. Yeah, whatever. For the record, I had exactly the same problem with exactly the same response. I left one message a day for several days with no response. When I e-mailed about it I was told that that was impossible: I couldn't have called, the number was not on the webpage. I then showed him the link. He said the information wasn't there. When I went back, he was right: it wasn't there. Funny, though, the Google cache still showed the phone number... I can't speak about NuFone's service. I never got that far. I'm now using a different VoIP provider. However, when a person tells me what I have done was *impossible*, and when I show him how I did it he *still* doesn't follow up, I don't need to do business with them. Obviously, it's still a problem. You would think that when multiple people tell you that they have tried to get in touch with you and the messages are falling on the floor, you would do something about it. I guess they're too busy improving their service to take on additional customers... I've held off writing this to the list, but this was just too much. Mr. McNamara replied to Mr. Baker with the same brisque statement he made to me several months ago. I found his reaction unpleasant then, and I find it disturbing that others are having the same problem months later. Tim Massey ___ 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
Re: [Asterisk-Users] Zapateller
If you want to test it, once you include Zapateller in your dial plan, place an incoming and block your caller id (from the phone your testing from) and it will do its job.. to block incoming calls that dont produce a caller id. J.C. - Original Message - From: Chris Albertson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 27, 2004 1:49 PM Subject: Re: [Asterisk-Users] Zapateller I have a few (more specific) questions about 'Zapateller': 1) How would you test this?? Would I need a predictive dialer machine like the telemarketers use. OK, I could just wait and see if it seems to cut down the unwanted calls but that's not really a test. 2) I don't understand how I would choose to use the answer option or not. under what conditions would answer by better then not using the option? I asume if I used answer the tones would play in caller's ear and be _very_ anoying to them. Why would I want to do that? There must be some reason are why would the option exist? --- Steve Foy [EMAIL PROTECTED] wrote: Hi, I'm just wondering about 'Zapateller'. How exactly does it work!? I might be interested in employing it at work here, but wondering if anyone's using it? echo*CLI show application Zapateller -= Info about application 'Zapateller' =- [Synopsis]: Block telemarketers with SIT [Description]: Zapateller(options): Generates special information tone to block telemarketers from calling you. Returns 0 normally or -1 on hangup. Options is a pipe-delimited list of options. The following options are available: 'answer' causes the line to be answered before playing the tone, 'nocallerid' causes Zapateller to only play the tone if there is no callerid information available. Options should be separated by | characters -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ 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 ___ 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
[Asterisk-Users] Does anyone manage the wiki?
I would like to correct some of the text on the GotoIf application page on the wiki. Does somebody actively manage changes like this, or should I fire away and make it myself? I'm actually surprised I have permission to edit a page without prior authorization, but it DOES state at the bottom of the generated pages to 'please update the page with new information...'. If I don't hear otherwise, I'll go ahead. -- Chris ___ 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
Re: [Asterisk-Users] Test
That is no longer a problem. bkw On Wed, 28 Jan 2004, Tazman wrote: Hi, a Virus was sent from this account to the Asterisk-Users mailing list... scan your computer for virus!!! Ronen - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 26, 2004 7:29 PM Subject: [Asterisk-Users] Test The message contains Unicode characters and has been sent as a binary attachment. ___ 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 ___ 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
[Asterisk-Users] SIP error
Hello When ever i make calls via a SIP provider I keep getting this error message Jan 29 02:09:20 NOTICE[1228887360]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible any idea what is it? Regards Deepak
RE: [Asterisk-Users] Asterisk Appliance
We have done this with nexcom.com appliances. We are currently using EBS 1569PS unit with a 1gig flash card. Keep in mind were only using the gateway features of asterisk on this box. If you are using the full PBX features you may want to use a more powerful model. John Bittner Simlab.net Nexcom contact info. Brian D. Earnhart Regional Sales Manager Nex Computer Inc. 46707 Fremont Blvd. Fremont CA 94538 510-656-2248 x 13 510-656-2158 Fax 510-396-7753 Mobile [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Martin Sent: Wednesday, January 28, 2004 12:08 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Appliance Ok now that we have a Asterisk server running quite well, we want to put it onto a more appropriate device, i.e. not a beige box computer, but perhaps some kind of embedded linux appliance. Has anyone already done this? Any suggestions on some tidy, small, suitable linux systems to use for asterisk? I.e., somthing that looks like this: https://secure.makonetworks.com//images/main/mako_250_shad.gif ___ 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
[Asterisk-Users] test
test ___ 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
[Asterisk-Users] Channels
Hi, I have a TE410P card anda configuration with IAX (soft phone). How * is using the chan_iax/zap ? What sequence and for what ? Also, at zap channels there is a kind of result recognition. Meaning, the busy tone, no answer etc. So, when recognizes a status, automatically creates a record (CDR) ? Why, there are not options for Fax and Anwer Machines ? Is there a solution or work around ? I think thats a major improvement or not ? Thanks, Marin Blu Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it!
Re: [Asterisk-Users] Junk calls from FWD numbers
On Tue, 27 Jan 2004, Chris Albertson wrote: Question: Does everyone with an FWD number get these junk calls or am I the only lucky one? I just got an FWD number a couple days ago, but haven't had that experience yet. And no, I haven't tried calling you to see if you'd answer. :) Greg ___ 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
Re: [Asterisk-Users] RE: Bluetooth discussions (quick glance to some BT products)
Ok I am looking at the Clipcomm solution. Bluetooth enabled phones can place voip calls by communicating with the BS-V100/L100? Am I getting this right? What is this feature called in phones and which cell phones support this? - Dustin - Eric Bart wrote: There are somes products available claiming to connect a BT headset, a cell phone and a phone land line all together. I've found some : http://www.geekzone.co.nz/content.asp?contentid=2079 http://www.clipcomm.co.kr/ The clipcomm BS-A101 sample price is : $570. It's VoIP land phone that can make/receive a voice call using your CTP cell phone or BT-enabled headset. Maybe the best system is: http://www1.norwoodsystems.com/ It's complete wireless solution that include hot connection to bluetooth hubs when the employee walks in the office. But it's not currently available. Maybe it'll work better with Bluetooth 1.2 wich will be very soon on the market. Here's a reply I got from norwoodsystems : Thanks for your enquiry. I attach a product description to help you understand our solution. We intend to sell the software through resellers and are currently reviewing the business model and packaging. We can achieve a price of under EUR450 per user including software, bluetooth headsets or phone upgrade to CTP enabled GSM phone( Bluetooth Cordless Telephony Profile), VoIP gateway, Bluetooth USB dongles, ISDN Basic Rate Interface, and software installation. ___ 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 ___ 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
RE: [Asterisk-Users] Does anyone manage the wiki?
Go ahead and edit the page. I've fixed several little errors on pages that I didn't create. The voip-info.org Wiki is like the total-open-source Asterisk manual. Although the total-access may be a problem in the future because all someone has to do to delete everything is just to register and start deleting. MATT--- -Original Message- From: Chris Higgins [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 28, 2004 10:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Does anyone manage the wiki? I would like to correct some of the text on the GotoIf application page on the wiki. Does somebody actively manage changes like this, or should I fire away and make it myself? I'm actually surprised I have permission to edit a page without prior authorization, but it DOES state at the bottom of the generated pages to 'please update the page with new information...'. If I don't hear otherwise, I'll go ahead. -- Chris ___ 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 ___ 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
Re: [Asterisk-Users] X-Lite Asterisk: Speex iLBC not working?
check out the latests cvs it has two reg files that will fix xlite or xpro to work bkw On Mon, 26 Jan 2004, Wim Venneman wrote: If this may be of any use: I'm not an expert but I did the test with the FWD soft phone from X-ten and iLBC SPX don't work. Asterisk wasn't between the connections. Just x-lite and fwd (who is an Asterisk Server?) The soft phone makes the connection but I can't hear any sound. Wim - Original Message - From: Fran Boon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 26, 2004 3:01 PM Subject: [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 forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing iLBC, there is some very garbled noise, but nothing intelligible. Sniffing the packets, I can see that X-Lite Asterisk have chosen differing 'Payload type' numbers: X-Lite: a=rtpmap:97 speex/8000 a=rtpmap:98 iLBC/8000 Asterisk: a=rtpmap:97 iLBC/8000 a=rtpmap:110 SPEEX/8000 According to the Speex RFC, this is acceptable: http://speex.org/drafts/draft-herlein-speex-rtp-profile-00.txt Dynamic payload type codes MUST be negotiated 'out-of-band' for the assignment of a dynamic payload type from the range of 96-127. I'm wondering whether the system is at all case sensitive? From the RFC: When conveying information by SDP [4], the encoding name SHALL be speex. NB Ethereal shows payload-type as being 97 when X-Lite reports iLBC 110 when X-Lite reports Speex, so the Asterisk numbers seem to 'win'. Any light shed on this would be great. Whilst GSM is ok, it would be great to leverage the power of Speex :) Thanks a lot, Fran. ___ 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 ___ 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 ___ 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
[Asterisk-Users] testing
testing yet again. bkw ___ 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
Re: [Asterisk-Users] Has Nufone gone belly-up
What you (and a lot of consumers) don't understand is that some customers need to get fired. Some people aren't worth doing business with. Those are the facts. I have done this. I have felt better after I fired a cusotmer in situations like that. bkw ___ 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
[Asterisk-Users] lists.digium.com
All of the mailing lists are now filtered for virues and spam before they reach digiums network. This will ensure that the mess that caused the list server to break down wont happen again. Its still playing catchup. But the mail is now flowing faster than usual now. Thanks, Brian (bkw_) ___ 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
