RE: [Asterisk-Users] Sangoma and Digium same machine?
I'm using a Sangoma A101 card alongside an older TDM400 and they seem to be playing nice. I've had it in production for a few months now with no problems. Thanks, Reid Forrest, CISSP Max-IS Inc. [EMAIL PROTECTED] Direct/Cell: 321-214- Main: 407-786-9600 > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of William Lloyd > Sent: Monday, September 26, 2005 1:08 PM > To: asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Sangoma and Digium same machine? > > Anybody ever put a Sangoma and a Digium card in the same server? > > Specifically a four port card from each company? > > -bill > [EMAIL PROTECTED] > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] AstriCon 2006 Location
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Brian Capouch > Sent: Sunday, September 18, 2005 12:32 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] AstriCon 2006 Location > > Senad J wrote: > >>If you are looking for the maximum number of cheap flights from around > >>the world, and plenty of convention and room space, the answer is Las > >>Vegas :-) > > > > > > I would definitively agree! > > > > Yes, but what would one do there? > > One who doesn't gamble, drink, or carouse, that is. > > I am making my first trip to LV later this Fall, and I dread it. I > can't imagine what I'll be able to find to do when I'm not at the > conference. > Have you been there before? I spent a week there a few years ago and found plenty to do outside the casinos. There are plenty of good (clean) shows, exhibits, and attractions. Even if you don't like to shop (I don't) the Forum Shops at Caesar's Palace are great. If you're a Star Trek fan, the Star Trek Experience is a must see. R ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] AstriCon 2006 Location
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Kevin Bockman > Sent: Saturday, September 17, 2005 2:39 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] AstriCon 2006 Location > > Matthew Simpson wrote: > > Atlanta is hub for Delta and Airtran > > > > Dallas is hub for American > > > > Chicago is hub for ATA > > > > All good central locations with cheap non stop flights. > > Atlanta is central for who? With all of the tornados, hurricanes, etc. > I would definately vote no for there. Dallas and Chicago are both good. > How about Orlando? Plenty to do here, and no lack of flights! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] TDM400P stops answering
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Kevin P. Fleming > Sent: Tuesday, September 13, 2005 6:13 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] TDM400P stops answering > > Andy Howell wrote: > > I have a weird problem in which my digium card stops answering. After > > running for a couple days, incoming calls are not seen. Running asterisk > > -r shows no incoming calls. Restarting Asterisk does not help. After a > > reboot it is fine. > > This problem was fixed in CVS (HEAD and v1-0) quite some time ago; what > versions are you running? I think this could also be a hardware issue. There have been several threads over the last year or so discussing older TDM400 cards and this very problem. We experience the same thing with a TDM400 card that's about a year and a half old. I've heard newer cards don't have this flaw. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Problem with Sangoma/Adtran 600 installation
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) > Sent: Saturday, April 30, 2005 5:59 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [Asterisk-Users] Problem with Sangoma/Adtran 600 installation > > I have installed Asterisk on a CentOS4 box and then installed Asterisk > from > CVS. > [EMAIL PROTECTED] asterisk]# ztcfg > Notice: Configuration file is /etc/zaptel.conf > line 0: Unable to open master device '/dev/zap/ctl' > > 1 error(s) detected > > When I run "service zaptel restart" I get: > > Waiting for zap to come online...Error: missing /dev/zap! > > Wha am I doing wrong? > [Reid Forrest] Check out README.udev in the zaptel source directory. You need to make modifications to the udev config files. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Group Extension
> > i need to create a group extension, to make calls to 6 sw > phones, but i > need to know if asterisk can do help me to get a unique > number and check > what extension has received less calls than the others, and > pass the new > call. We got a call center and want to know if we can distribute the > calls depending in what extension is available and from the extensions > that are available pass the call to the operator that has > answered less > calls, can i do this with *? can i get statistics from the use for an > extension? can anybody help me?? You're looking for ACD (automatic call distribution). Check the wiki for help: http://voip-info.org/tiki-index.php?page=Asterisk%20config%20queues.conf http://voip-info.org/tiki-index.php?page=Asterisk%20config%20agents.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Autio cut off at beginning of call
I'm running Asterisk 1.0.5 stable. Before that I have run 1.0.3, 1.0.1, 1.0 and many CVS versions all with the same symptoms. Thank you, Reid Forrest, CISSP Max-IS, Inc. [EMAIL PROTECTED] ofc: 407.786.9600 x1200 cell: 321.439.8903 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Keith O'BrienSent: Tuesday, January 25, 2005 8:24 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Re: Autio cut off at beginning of call For what it is worth I am experiencing the exact same problem with the latest CVS. I have tried numerous IAX providers and the problem follows so it isn’t the provider. Are you running stable or CVS? I think that you hit it on the head, the fact that audio is being sent prior to an IAX ANSWER seems like a bug to me. I verified that I am seeing this same symptom of audio packets before the answer packet. If no one disagrees I will open a bug. == > Check the load on your server(s). > Load average is always at or near 0. This is on a dedicated machine doingnothing but routing calls. No voicemail, music on hold, etc. I noticed something in a packet capture that may or may not be significant.When I place a call, the capture shows about 3 seconds of audio data beforethe IAX ANSWER packet. Could this be a symptom of the problem? Below is a packet dump if an outbound IAX call made from a Zaptel FXSchannel. No. Time Source Destination Protocol Info 9 11.987842 asterisk 66.234.228.170 IAX2 IAX,source call# 4, timestamp 3ms REGREQ 11 12.086365 66.234.228.170 asterisk IAX2 IAX,source call# 356, timestamp 4ms REGACK 12 12.086456 asterisk 66.234.228.170 IAX2 IAX,source call# 4, timestamp 4ms ACK 13 12.158422 asterisk 66.234.228.160 IAX2 IAX,source call# 6, timestamp 14ms NEW 14 12.258421 66.234.228.160 asterisk IAX2 IAX,source call# 228, timestamp 10ms AUTHREQ 15 12.258514 asterisk 66.234.228.160 IAX2 IAX,source call# 6, timestamp 114ms AUTHREP 16 12.383395 66.234.228.160 asterisk IAX2 IAX,source call# 228, timestamp 109ms ACCEPT 17 12.383552 asterisk 66.234.228.160 IAX2 IAX,source call# 6, timestamp 109ms ACK 18 12.397652 asterisk 66.234.228.160 IAX2 Voice,source call# 6, timestamp 253ms, Raw mu-law data (G.711) 19 12.417636 asterisk 66.234.228.160 IAX2 Minipacket, source call# 6, timestamp 273ms, Raw mu-law data (G.711) 20 12.437635 asterisk 66.234.228.160 IAX2 Minipacket, source call# 6, timestamp 293ms, Raw mu-law data (G.711) 21 12.457634 asterisk 66.234.228.160 IAX2 Minipacket, source call# 6, timestamp 313ms, Raw mu-law data (G.711) 22 12.477634 asterisk 66.234.228.160 IAX2 Minipacket, source call# 6, timestamp 333ms, Raw mu-law data (G.711) 23 12.497634 asterisk 66.234.228.160 IAX2 Minipacket, source call# 6, timestamp 353ms, Raw mu-law data (G.711) 24 12.498398 66.234.228.160 asterisk IAX2 IAX,source call# 228, timestamp 253ms ACK 25 12.515119 66.234.228.160 asterisk IAX2Control, source call# 228, timestamp 112ms stop sounds 26 12.515161 asterisk 66.234.228.160 IAX2 IAX,source call# 6, timestamp 112ms ACK 27 12.517701 asterisk 66.234.228.160 IAX2 Minipacket, source call# 6, timestamp 373ms, Raw mu-law data (G.711) 28 12.537634 asterisk 66.234.228.160 IAX2 Minipacket, source call# 6, timestamp 393ms, Raw mu-law data (G.711) 29 12.557634 asterisk 66.234.228.160 IAX2 Minipacket, source call# 6, timestamp 413ms, Raw mu-law data (G.711) 30 12.565685 66.234.228.160 asterisk IAX2Control, source call# 228, timestamp 115ms unknown (0x0e) 31 12.565732 asterisk 66.234.228.160 IAX2 IAX,source call# 6, timestamp 115ms ACK 32 12.577636 asterisk 66.234.228.160 IAX2 Minipacket, source call# 6, timestamp 433ms, Raw mu-law data (G.711) 33 12.585428 66.234.228.160 asterisk IAX2 Voice,source call# 228, timestamp 20ms, Raw mu-law data (G.711) 34 12.585505 asterisk 66.234.228.160 IAX2 IAX,source call# 6, timestamp 20ms ACK 35 12.597633 asterisk 66.234.228.160 IAX2 Minipacket, source call# 6, timestamp 453ms, Raw mu-law data (G.711) 36 12.606341 66.234.228.160 asterisk IAX2 Minipacket, source call# 228, timestamp 40ms, Raw mu-law data (G.711) 37 12.617633 asterisk 66.234.228.160 IAX2 Minipacket, source call# 6, timestamp 4
RE: [Asterisk-Users] Autio cut off at beginning of call
> > Check the load on your server(s). > Load average is always at or near 0. This is on a dedicated machine doing nothing but routing calls. No voicemail, music on hold, etc. I noticed something in a packet capture that may or may not be significant. When I place a call, the capture shows about 3 seconds of audio data before the IAX ANSWER packet. Could this be a symptom of the problem? Below is a packet dump if an outbound IAX call made from a Zaptel FXS channel. No. TimeSourceDestination Protocol Info 9 11.987842 asterisk 66.234.228.170IAX2 IAX, source call# 4, timestamp 3ms REGREQ 11 12.086365 66.234.228.170asterisk IAX2 IAX, source call# 356, timestamp 4ms REGACK 12 12.086456 asterisk 66.234.228.170IAX2 IAX, source call# 4, timestamp 4ms ACK 13 12.158422 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 14ms NEW 14 12.258421 66.234.228.160asterisk IAX2 IAX, source call# 228, timestamp 10ms AUTHREQ 15 12.258514 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 114ms AUTHREP 16 12.383395 66.234.228.160asterisk IAX2 IAX, source call# 228, timestamp 109ms ACCEPT 17 12.383552 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 109ms ACK 18 12.397652 asterisk 66.234.228.160IAX2 Voice, source call# 6, timestamp 253ms, Raw mu-law data (G.711) 19 12.417636 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 273ms, Raw mu-law data (G.711) 20 12.437635 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 293ms, Raw mu-law data (G.711) 21 12.457634 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 313ms, Raw mu-law data (G.711) 22 12.477634 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 333ms, Raw mu-law data (G.711) 23 12.497634 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 353ms, Raw mu-law data (G.711) 24 12.498398 66.234.228.160asterisk IAX2 IAX, source call# 228, timestamp 253ms ACK 25 12.515119 66.234.228.160asterisk IAX2 Control, source call# 228, timestamp 112ms stop sounds 26 12.515161 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 112ms ACK 27 12.517701 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 373ms, Raw mu-law data (G.711) 28 12.537634 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 393ms, Raw mu-law data (G.711) 29 12.557634 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 413ms, Raw mu-law data (G.711) 30 12.565685 66.234.228.160asterisk IAX2 Control, source call# 228, timestamp 115ms unknown (0x0e) 31 12.565732 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 115ms ACK 32 12.577636 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 433ms, Raw mu-law data (G.711) 33 12.585428 66.234.228.160asterisk IAX2 Voice, source call# 228, timestamp 20ms, Raw mu-law data (G.711) 34 12.585505 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 20ms ACK 35 12.597633 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 453ms, Raw mu-law data (G.711) 36 12.606341 66.234.228.160asterisk IAX2 Mini packet, source call# 228, timestamp 40ms, Raw mu-law data (G.711) 37 12.617633 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 473ms, Raw mu-law data (G.711) . . . Nothing of interest in here, just audio data . . 327 15.477620 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp ms, Raw mu-law data (G.711) 328 15.486797 66.234.228.160asterisk IAX2 Mini packet, source call# 228, timestamp 2920ms, Raw mu-law data (G.711) --->329 15.487713 66.234.228.160asterisk IAX2 Control, source call# 228, timestamp 2923ms ANSWER 330 15.487734 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 2923ms ACK 331 15.497627 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 3353ms, Raw mu-law data (G.711) 332 15.517623 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 3373ms, Raw mu-law data (G.711) 3
RE: [Asterisk-Users] Autio cut off at beginning of call
> > This is not normal; I do *not* have this issue with NuFone > and I have placed a > ton of calls through them daily for the past year. I don't > recall having > this problem with voicepulse connect when I used them, nor do > I have the > issue with iax.cc for inbound calls. > I'm experiencing this on two separate * systems. The symptoms appear only on outbound calls, never inbound. I think it's important to note that this affects outbound calls made either of SIP or IAX, and through multiple providers. > It very much sounds like it's something on your end... How about some > specifics? I'm using the defaults found in the [general] section of iax.conf and sip.conf. I'm using Asterisk version 1.0.3, but I've experienced this problem with every version I've used over the past year. It also does not matter if the call is placed from a SIP phone or an FXS channel. What additional info would be most helpful? Thanks, Reid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Autio cut off at beginning of call
I posted this question a while back, and I'm posting again in hopes that someone has some ideas. Sorry if you've already seen this. When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom, VoicePulse Connect) I often find that after the call is answered the first few seconds of audio are cut off (i.e. I don't hear the called party). This usually results in the called party saying "hello Hello???" until I hear them. Has anyone else experienced this problem and found a cause or fix? My internal calls are perfect. It's just Internet-terminated calls that have the problem. Someone wrote in response to the last post saying that the audio path probably wasn't set up yet. I think this is the symptom, but I'm wondering what's the cause, and if there's a fix. Surely I'm not the only one who's having a huge problem with this. Can anyone help? Thank you, Reid Forrest, CISSP Max-IS, Inc. [EMAIL PROTECTED] ofc: 407.786.9600 x1200 cell: 321.439.8903 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clipping on outbound calls via SIP/IAX
I'm hoping someone can help me with a problem I've been having for a while now. I've googled and wiki'd to no avail. Whenever I place an outbound call from * to a PSTN through a SIP or IAX provider (e.g. Voicepulse or Broadvoice), the first 1/2 to 2 seconds of the remote call are clipped (muted). For example, if I call a remote voicemail system that usually answers with "Nortel Call Pilot, Mailbox?" I might get "ilot, Mailbox?". Everything works fine if I dial an internal extension or through the PSTN. Is this just something I'm going to have to live with if using an Internet-based termination provider? I'm using Asterisk 1.0.3 and have tested on different systems, different providers, different phones, etc. Thank you, Reid Forrest, CISSP Max-IS, Inc. [EMAIL PROTECTED] ofc: 407.786.9600 x1200 cell: 321.439.8903 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] dialplan selection
>
> [globals]
> X1000=SIP/1000
> X1001=ZAP/1001
> X1002=IAX2/1002
> X1003=SIP/1003
>
> [outbound]
> exten => _123,1,Dial(${X${EXTEN:4}},10)
>
Oops, that line should read:
exten => _123,1,Dial(${X${EXTEN:3}},10)
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RE: [Asterisk-Users] dialplan selection
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Samudra E. Haque
> Sent: Sunday, December 19, 2004 12:58 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] dialplan selection
>
> Hello,
>
> I would like to parse inbound Asterisk IAX2 7-digit numbers
> in the form of
> 123-4567 and strip out the first four digits, and then dial
> whatever number
> digits remain. If I only have three digits (000-999) and have a mix of
> channels (ZAP, SIP, IAX2) could someone please point out how
> I can use a
> single DIAL command to just dial the extension regardless of
> the type of
> channel. .. For each valid extension, I have a separate dial
> command anyway,
> which denotes the particular channel that extension is assigned to.
>
> I do not want to assign groups of extensions i.e., 123-A567
> or 123-B567 or
> 123-C567 where A=ZAP, B=SIP, C=IAX2 peers respectively.
>
Samudra,
If I understand you correctly, you're not just looking to strip digits, but
dial an arbitrary extension without specifying the channel type in the dial
command. correct?
You should be able to accomplish this using a variable for each extension.
For example:
[globals]
X1000=SIP/1000
X1001=ZAP/1001
X1002=IAX2/1002
X1003=SIP/1003
[outbound]
exten => _123,1,Dial(${X${EXTEN:4}},10)
If the user dials 1231002, then Dial(IAX2/1002,10) should be executed.
If you have a lot of extensions then you should be able to put the variables
into a database instead.
Reid
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RE: [Asterisk-Users] CPC, Calling Party Control, Disconnect supervision, -- how to tell that to Verizon (east coast)?
> > It's an analog POTS line connected to a TDM400 interface in > the * box. Config > > files are set for kewlstart. > > I've never heard of this problem. Analog FXO ports on Asterisk are > considered answered when Asterisk finishes sending the DTMF. > > Do you have callprogress=yes in /etc/asterisk/zapata.conf? > If so, set > it to "no". > That did the trick. Thanks for the tip! ___ 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] CPC, Calling Party Control, Disconnect supervision, -- how to tell that to Verizon (east coast)?
> > On the surface, that sounds like an * problem, not sprint. > > What are you using to interface to sprint (analog, bri, T1), which > cards in your * box (and associated config files)? > > A fairly standard telco operating approach is _not_ to > provide any answer > supervision, and * works just fine without it for lots of folks. > It's an analog POTS line connected to a TDM400 interface in the * box. Config files are set for kewlstart. ___ 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] CPC, Calling Party Control, Disconnect supervision, -- how to tell that to Verizon (east coast)?
> > > > I would be grateful if anybody could tell me what I should > > tell Verizon > > in NJ so they would enable "disconnect supervision" for my lines. > > > > Apparently "remote hangup notification" or "disconnect > supervision" or > > "calling party control" is NOT the magic phrase for them. Although > > disconnect supervision is in the glossary on their website. This may be off topic for this post, but I'm having a similar issue with Sprint. I finally did get disconnect supervision turned on (for an extra $5/month), but don't have answer supervision. Even though the remote party answers and we can converse, Asterisk doesn't know the call was answered and will time out after a short period. Any suggestions? Sprint has now closed out three tickets with no resolution. ___ 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] Kind of off-topic: VoIP services and multiplecallers
> -Original Message- > service is compatible with Asterisk). However, I have a question: can > more than one person make/receive a call at the same using one VoIP > line? > > If five people in the office all need to use their phones at the same > time, would I need five VoIP lines, or would I only need one > VoIP line? > Am I over-thinking this? No, you're not over-thinking at all. The answer is that it depends on the provider. The two that I have firsthand experience with are Voicepulse Connect and Broadvoice. VP Connect will allow at least four simultaneous inbound or outbound calls. Since they bill per minute on all outbound calls, the rate is the same no matter how many calls are active. I've heard that they've bumped up the limit to more than 4, but I haven't tried it myself. Broadvoice will also allow multiple simultaneous calls. The first outbound call is included in your plan. Additional outbound calls cost $0.039 / minute each. Although I haven't used them myself, I understand that VoipJet allow multiple outbound calls within the US for $0.013 / minute. Thank you, Reid Forrest, CISSP Max-IS, Inc. [EMAIL PROTECTED] ofc: 407.786.9600 x1200 cell: 321.439.8903 ___ 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] Broadvoice outbound 404 error
Is anyone else experiencing 404 errors on outbound dial with Broadvoice? I've followed the instructions posted by Broadvoice to configure sip.conf, and inbound calling works fine. Every time I try to dial out, I get a 404 "Not Found" error. Here are the relevant sections from my configs. sip.conf: context=broadvoice-in register => [EMAIL PROTECTED]:xxpasswordxx:[EMAIL PROTECTED] [bv-home] type=peer host=proxy.dca.broadvoice.com fromdomain=sip.broadvoice.com fromuser=3215551212 context=inbound canreinvite=no qualify=yes disallow=all allow=ilbc allow=gsm allow=ulaw dtmfmode=inband secret=xxpasswordxx insecure=very Thank you, Reid Forrest, CISSP Max-IS, Inc. [EMAIL PROTECTED] ofc: 407.786.9600 x1200 cell: 321.439.8903 ___ 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
