Re: [Asterisk-Users] Unable to detect process 256 frames
Does the same apply to GSM channels? - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, December 19, 2003 9:34 PM Subject: Re: [Asterisk-Users] Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames Do not try to do inband DTMF on G.729 Can we wiki-fy this? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 docs
Title: Chan_h323 docs Jeremy, In some posting in the mailing lists, you mentioned that docs for h323 had been submitted but hadn't made it into distribution. Could you post those docs in your download directory? I'm trying to understand the nuances of your driver, gnugk, and extensions. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
[Asterisk-Users] ZTMonitor - /dev/dsp problem
Hi, I am trying to run ZTMonitor to get debug info from my E100P board but I got the following message: -bash-2.05b# ./ztmonitor 1 Unable to open /dev/dsp: No such file or directory Cannot open audio ... -bash-2.05b# Thanks, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] SNOM 200 and * issues
We included presence in the latest builds. This is necessary to enable auto-redial (call completion), at least in the eyes of the SIP hot-shots. In the latest image (2.03c, see http://snom.com/download/share) we included a flag where this feature can be turned off. But as long as it's not causing any harm, you should just ignore this. Christian -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Michael Graves Gesendet: Samstag, 20. Dezember 2003 00:38 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] SNOM 200 and * issues Hello All, My SNOM 200 phone keeps generating the following message on the * console: Notice [11127005368] : File chan_sip.c Line 5394 (handle_request) : Unknown sip command 'Publish' from '192.168.1.17' What does this mean and how do I remedy the problem? Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 One day in your life shouldn't be a problem - 54-40 from One Day ** 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk MGCP register
Hi, I am trying to figure out if * can register as a client on a remote MGCP service. Just like SIP and other protocols Do. Anyone tried this? Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 gnugk
Ok, I've managed to get inbound and outbound calling to work with chan_h323 and gnugk. A few questions: 1) if I do a reload in *, chan_h323 loses its registration with gnugk, and will no longer pass calls to it. A second reload will crash *. Is this supposed to be? 2) For a configuration in h323.conf like: [office] type=h323 prefix=9 context=outbound I get a message saying: WARNING[1074403072]: File chan_h323.c, Line 215 (build_alias): Keyword h323 does not make sense in type=h323 Why is that? 3) When making a call from an h323 client such as ohPhone registered with gnugk, I can make the call, but it uses the context from the [general] section, rather than the context in [office]. Is this supposed to be? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best SIP PHones to buy ?
Hi People, Can anyone help-me here with a simple question. I wanna buy a Sip Phone, but what is the best and cheap one ? I see alot of messages about, grandstream , snow etc etc. So for use with my * system, what sip phone is the best ?? Can him be used behind a nat system etc ? Or with a Broadband connection etc ? Thanks alot for help!! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR sample config?
Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes, I did look at the wiki and google searched for ivr menu.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IVR sample config?
Rich- Chapter 4 of the (so-called) draft handbook, details what you need to know pretty well. Here's the link: http://www.digium.com/handbook-draft.pdf Regards, Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Saturday, December 20, 2003 2:37 PM To: Asterisk-a-users-list Subject: [Asterisk-Users] IVR sample config? Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes, I did look at the wiki and google searched for ivr menu.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX phone busy
This was a typo--it already is: exten = 70,1,Dial(IAX/mike,30,tr) and I still get a busy... Thoughts? Dan wrote: Hi Michael, - Original Message - From: Michael Welter [EMAIL PROTECTED] Subject: [Asterisk-Users] DIAX phone busy I've configured the DIAX phone. It registers with the * server, and I am able to make calls from DIAX. Did you registered using IAX or IAX2? Check if in Registering page you have selected IAX2 or not. However, when I try to call the DIAX phone from another phone, I get a busy signal. My extensions.conf: exten = 70,1,Dial(IAX/mike/mike,30,tr) # IAX Mike This is wrong. You must have: exten = 70,1,Dial(IAX/mike,30,tr) or better, in order to support both IAX and IAX2 type of registrations: exten = 70,1,Dial(IAX/mikeIAX2/mike,30,tr) Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IVR sample config?
exten = _X., Goto(ivrmenu,s,1) [ivrmenu] exten = s,1,Ringing exten = s,2,DigitTimeout,30 exten = s,3,Background(something) ; press 1 for sales exten = t,3,Goto(business,0,1) include = business ;map yourl extens here... [business] exten = 0,1,... exten = 1,1, .. From: Rich Adamson [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: Asterisk-a-users-list [EMAIL PROTECTED] Subject: [Asterisk-Users] IVR sample config? Date: Sat, 20 Dec 2003 08:37:26 -0600 Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes, I did look at the wiki and google searched for ivr menu.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Dont miss out on jobs that are not advertised. http://go.msnserver.com/IN/38902.asp Post your CV on naukri.com today. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Crash
I didn't repeat this question. I read the responses and amended my request. I was simply asking for any suggestions as how to diagnose this problem including the ways to provide additional information to this list to help solve the problem. After reinstalling Linux and Asterisk from scratch with no improvement, a helpful member of this list suggested it was the way I was starting asterisk from the command line. Started asterisk from a script and the problem is resolved. I provide this update to the group with the hopes that it helps anybody with a similar issue. Not every one is an expert and I am thankful for the assistance. -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Thursday, December 18, 2003 12:59 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Crash On Thu, 2003-12-18 at 07:25, Kevin wrote: Asterisk Crash I have an application that using the System() command. When ever I invoke the command my asterisk crashes. I have updated to the latest CVS and it crashes. Can someone offer some suggestions on how to diagnose and correct this problem? Thanks Kevin Extensions.conf exten = 2810,1,System(date) exten = 2810,2,Goodbye repeating your question with no additional information will likely have it ignored. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iconnect 480 unavailable msgs
Hi guys i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few problems. Im running asterisk and xp/xlite softphone.. both xp box and asterisk box are on public ips. The problem is that when i ring anyone in the world it'll ring they'll pickup and i can hear them 100% perfectly/clearly.. but they cant hear me.. occasionaly they can hear something like a tiny bit of a word once in a while. im getting the error msgs back Got SIP response 480 Temporarily not available back from 213.137.73.176.. i had a look on google and aparantly if you change the codec it fixed it or something.. i tried all of them nothing seemed to work. heres a copy of my sip.conf [iconnect] type=friend username= secret= host=sipauth.deltathree.com qualify=1000 callerid=178197026 allow=g721.1 and for the xlite [2001] type=friend callerid=JUSTIN XLITE 2001 username=2001 secret=2001 context=-phones mailbox=2001 host=dynamic nat=no qualify=yes allow=9721.1 ive been playing around with the nat settings to just incase... alltho neither box's are using nat. Any oneknow how i go about fixing this ? would be cool if ppl could hear me to on the phone heh :) thanks heaps ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX phone busy
Hi, Did you registered using IAX or IAX2? Check if in Registering page you have selected IAX2 or not. Did you check this one too? BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect 480 unavailable msgs
Hi, try to put in both users setup, canreinvite = no and what is this codec allow=g721.1 ... :))) Lubo vocalvoip wrote: Hi guys i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few problems. Im running asterisk and xp/xlite softphone.. both xp box and asterisk box are on public ips. The problem is that when i ring anyone in the world it'll ring they'll pickup and i can hear them 100% perfectly/clearly.. but they cant hear me.. occasionaly they can hear something like a tiny bit of a word once in a while. im getting the error msgs back Got SIP response 480 Temporarily not available back from 213.137.73.176.. i had a look on google and aparantly if you change the codec it fixed it or something.. i tried all of them nothing seemed to work. heres a copy of my sip.conf [iconnect] type=friend username= secret= host=sipauth.deltathree.com qualify=1000 callerid=178197026 allow=g721.1 and for the xlite [2001] type=friend callerid=JUSTIN XLITE 2001 username=2001 secret=2001 context=-phones mailbox=2001 host=dynamic nat=no qualify=yes allow=9721.1 ive been playing around with the nat settings to just incase... alltho neither box's are using nat. Any oneknow how i go about fixing this ? would be cool if ppl could hear me to on the phone heh :) thanks heaps ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IVR sample config?
Rich,
This is in response to your question about and IVR Menu. Below is the
dial-plan from the * at SAI. If you dial 1-800-747-9111 and the
Extension 2998, you'll be able to hear this one in action.
The key to creating it is to use Extension 205 (defined below) to record
your menu prompts. This will put the sound file in
/tmp/asterisk-recording.gsm. You'll have to move that file each time
its created to /var/lib/asterisk/sounds and rename it to something
pertinent to your design so it can be called from the dial-plan. Notice
the line under [mainmenu] exten = s,5,Background(sai-welcome). The
sai-welcome is one of those .gsm sound files. The rest of the dial-plan
just defines what happens when each option is pushed.
Hope this helps.
Joe
** extensions.conf ***
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5,Background(sai-welcome)
exten = s,6,Background(sai-choose)
; Tech Support
exten = 1,1,AGI(dima-test.agi)
exten = 1,2,SetGlobalVar(ACCOUNTCODE=${callerid})
exten = 1,3,SetVar(testcallerid=${callerid})
exten = 1,4,Background(sai-reptech-welcome)
exten = 1,5,Queue(rep-tech)
; Leave Voicemail
exten = 2,1,VoicemailMain()
exten = 2,2,Hangup
; Echo Test
exten = 3,1,Playback(demo-echotest)
exten = 3,2,Echo
exten = 3,3,Playback(demo-echodone)
exten = 3,4,Goto(mainmenu,s,6)
; EAGI Test
exten = 4,1,Answer()
exten = 4,2,Wait(1)
exten = 4,3,AGI(sai-repid.agi)
exten = 4,4,Wait(1)
exten = 4,5,Hangup
; Play Music-on-Hold
exten = 5,1,MusicOnHold(default)
exten = 5,2,Goto(mainmenu,s,6)
exten = #,1,Playback(sai-thanks)
exten = #,2,Hangup
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
[default]
include = mainmenu
include = local
include = longdistance
include = joe-iax
include = npi-iax
exten = 205,1,Wait(2) ; Call 205 to Record new Sound Files
exten = 205,2,Record(/tmp/asterisk-recording:gsm)
exten = 205,3,Wait(2)
exten = 205,4,Playback(/tmp/asterisk-recording)
exten = 205,5,wait(2)
exten = 205,6,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Saturday, December 20, 2003 8:37 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] IVR sample config?
Can someone point me to some reasonable example / starting point to
implement a basic IVR menu? Looking for something rather simple like the
press 1 for sales, 2 for tech support, and probably an option to list
the voicemail directory kind of thing. Nothing elaborate needed, just
basic menu.
(Yes, I did look at the wiki and google searched for ivr menu.)
___
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RE: [Asterisk-Users] iconnect 480 unavailable msgs
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of vocalvoip Sent: 20 December 2003 16:14 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] iconnect 480 unavailable msgs Hi guys i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few problems. Im running asterisk and xp/xlite softphone.. both xp box and asterisk box are on public ips. The problem is that when i ring anyone in the world it'll ring they'll pickup and i can hear them 100% perfectly/clearly.. but they cant hear me.. occasionaly they can hear something like a tiny bit of a word once in a while. im getting the error msgs back Got SIP response 480 Temporarily not available back from 213.137.73.176.. i had a look on google and aparantly if you change the codec it fixed it or something.. i tried all of them nothing seemed to work. heres a copy of my sip.conf [iconnect] type=friend username= secret= host=sipauth.deltathree.com qualify=1000 callerid=178197026 allow=g721.1 and for the xlite [2001] type=friend callerid=JUSTIN XLITE 2001 username=2001 secret=2001 context=-phones Should this have the - in the before phones? mailbox=2001 host=dynamic nat=no qualify=yes allow=9721.1Should this not be g721.1? Or should it be G723.1 ive been playing around with the nat settings to just incase... alltho neither box's are using nat. Any oneknow how i go about fixing this ? would be cool if ppl could hear me to on the phone heh :) thanks heaps ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect 480 unavailable msgs
[iconnect] type=friend username= secret= host=sipauth.deltathree.com qualify=1000 callerid=178197026 allow=g721.1 Hello...looks like you have your codecs messed up. If you have enough bandwidth I recommend you modify it to: [iconnect] type=friend username= secret= host=sipauth.deltathree.com disallow=all allow=ulaw and for the xlite [2001] type=friend callerid=JUSTIN XLITE 2001 username=2001 secret=2001 context=-phones Should this have the - in the before phones? mailbox=2001 host=dynamic nat=no qualify=yes allow=9721.1 Should this not be g721.1? Or should it be G723.1 same here: disallow=all allow=ulaw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x100P incoming
On Friday 19 December 2003 08:12, David Gomillion wrote: How do I make x100P does not answer incoming calls ? The only thing that springs to mind is that you create an incoming context, and have an extension like: Exten = s,1,Wait(1000) Dunno if it will work or not, but that's the only thing that springs to mind. What about setting in zapata.conf: immediate=no before the channel declaration for the X100P. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best SIP PHones to buy ?
On Sat, 20 Dec 2003, Carlos Arnt wrote: I wanna buy a Sip Phone, but what is the best and cheap one ? I see alot of messages about, grandstream , snow etc etc. We have bought around 30 Grandstream phones, both BT101 and BT102. In general the phone is reasonable, but it does have limitations. Notable issues for users tend to be the lack of any sort of consultative transfer or easy access to conference calling. Also its 'call waiting' facility is a rather annoying and loud normal ringing noise, rather than the usual 'beep beep' that people are probably used to and you can't disable the call waiting feature. Entering numbers also has problems as if you dial too quickly you tend to lose digits, even if you heard the tones and saw them appear on the display. We just bought a Snom200 which is a lot more expensive, but you do get what you pay for. It has a better display which copes with caller name as well as caller number, has consultative transfer, conference calling, no problems with losing digits when dialling and a normal 'call waiting' beep. It also has a host of other features, but we haven't used it long enough to find any problems as yet. I don't know what the Snom105 is like, but if it is anywhere near as good as the Snom200 it may be a good alternative to the BT101/2. Can him be used behind a nat system etc ? Both the Snom200 and BT101/2 have settings for use behind NAT. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More beginner questions
Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to the asterisk box. This seems to be quite useful software but it's frustratingly difficult to get running. Jon SIP debug shows following mrpenguin*CLI -- Registered '2203' (AUTHENTICATED) at 192.168.0.2:5036 -- Accepting AUTHENTICATED call from 192.168.0.2, requested format = 2, actu al format = 2 -- Executing SetCallerID([EMAIL PROTECTED]/9, 91184) in new stack -- Executing SetCIDName([EMAIL PROTECTED]/9, calisto) in new stack -- Executing Dial([EMAIL PROTECTED]/9, SIP/[EMAIL PROTECTED]) in new stack We're at 82.38.193.149 port 16612 Answering with preferred capability 4 Answering with preferred capability 2 Answering with non-codec capability 1 11 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 214 v=0 o=root 15141 15141 IN IP4 82.38.193.149 s=session c=IN IP4 82.38.193.149 t=0 0 m=audio 16612 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (NAT) to 192.246.69.223:5060 -- Called [EMAIL PROTECTED] Sip read: CLI SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: Free World Dialup (0.8.11rc3 (i386/linux)) Content-Length: 0 8 headers, 0 lines Sip read: CLI SIP/2.0 180 Ringing Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED];tag=as63b4567c Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Contact: sip:[EMAIL PROTECTED]:5028 Content-Length: 0 9 headers, 0 lines -- SIP/fwd.pulver.com-43fd is ringing Sip read: CLI SIP/2.0 180 Ringing Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED];tag=as2046b5cb Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 10 headers, 0 lines -- SIP/fwd.pulver.com-43fd is ringing Sip read: CLI SIP/2.0 200 OK Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 Record-Route: sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED];tag=as2046b5cb Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 212 v=0 o=root 11472 11472 IN IP4 65.121.72.14 s=session c=IN IP4 65.121.72.14 t=0 0 m=audio 12268 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 12 headers, 10 lines Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format GSM Found description format PCMU Found description format telephone-event Capabilities: us - 6, them - 6/0, combined - 6 Non-codec capabilities: us - 1, them - 1, combined - 1 list_route: hop: sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on list_route: hop: sip:[EMAIL PROTECTED] set_destination: Parsing sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on for addr ess/port to send to set_destination: set destination to 192.246.69.223, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 Route: sip:[EMAIL PROTECTED] From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED];tag=as2046b5cb Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 192.246.69.223:5060 -- SIP/fwd.pulver.com-43fd answered [EMAIL PROTECTED]/9 set_destination: Parsing sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on for addr ess/port to send to set_destination: set destination to 192.246.69.223, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 Route: sip:[EMAIL PROTECTED] From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED];tag=as2046b5cb Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent:
Re: [Asterisk-Users] x100P incoming
Exten = s,1,Wait(1000) This will make * not answering the call, but still you would see notices coming on your screen and also an entry in CDR. immediate=no Will try this out SW From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] x100P incoming Date: Sat, 20 Dec 2003 11:10:26 -0600 Reply-To: [EMAIL PROTECTED] On Friday 19 December 2003 08:12, David Gomillion wrote: How do I make x100P does not answer incoming calls ? The only thing that springs to mind is that you create an incoming context, and have an extension like: Exten = s,1,Wait(1000) Dunno if it will work or not, but that's the only thing that springs to mind. What about setting in zapata.conf: immediate=no before the channel declaration for the X100P. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] More beginner questions
Once the connection is made between your * and FWD, your * passes it off so its really just a connection between your soft-phone and FWD. So, your soft-phone needs to be properly NATed, and the appropriate ports TO your soft-phone need to be opened. Each SIP phone will attempt to use different ports (usually UDP), so you need to find out what those ports are and make sure they are opened and NATed to your soft-phone. I've never gotten it to work because I wasn't willing to open up my soft-phone that much to the Internet. I have experienced, however, a great deal of luck using IAX to communicate between two (actually three) * systems in different locations. The only port that needs to be opened is UDP 5036 for IAX, or UDP 4569 for IAX2. Good luck! Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Creasey Sent: Saturday, December 20, 2003 12:00 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] More beginner questions Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to the asterisk box. This seems to be quite useful software but it's frustratingly difficult to get running. Jon SIP debug shows following mrpenguin*CLI -- Registered '2203' (AUTHENTICATED) at 192.168.0.2:5036 -- Accepting AUTHENTICATED call from 192.168.0.2, requested format = 2, actu al format = 2 -- Executing SetCallerID([EMAIL PROTECTED]/9, 91184) in new stack -- Executing SetCIDName([EMAIL PROTECTED]/9, calisto) in new stack -- Executing Dial([EMAIL PROTECTED]/9, SIP/[EMAIL PROTECTED]) in new stack We're at 82.38.193.149 port 16612 Answering with preferred capability 4 Answering with preferred capability 2 Answering with non-codec capability 1 11 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 214 v=0 o=root 15141 15141 IN IP4 82.38.193.149 s=session c=IN IP4 82.38.193.149 t=0 0 m=audio 16612 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (NAT) to 192.246.69.223:5060 -- Called [EMAIL PROTECTED] Sip read: CLI SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: Free World Dialup (0.8.11rc3 (i386/linux)) Content-Length: 0 8 headers, 0 lines Sip read: CLI SIP/2.0 180 Ringing Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED];tag=as63b4567c Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Contact: sip:[EMAIL PROTECTED]:5028 Content-Length: 0 9 headers, 0 lines -- SIP/fwd.pulver.com-43fd is ringing Sip read: CLI SIP/2.0 180 Ringing Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED];tag=as2046b5cb Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 10 headers, 0 lines -- SIP/fwd.pulver.com-43fd is ringing Sip read: CLI SIP/2.0 200 OK Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 Record-Route: sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on From: calisto sip:[EMAIL PROTECTED];tag=as1f0e4544 To: sip:[EMAIL PROTECTED];tag=as2046b5cb Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 212 v=0 o=root 11472 11472 IN IP4 65.121.72.14 s=session c=IN IP4 65.121.72.14 t=0 0 m=audio 12268 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 12 headers, 10 lines Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format GSM Found description format PCMU Found description format telephone-event Capabilities: us - 6, them - 6/0, combined - 6 Non-codec capabilities: us - 1, them - 1, combined - 1 list_route: hop: sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on list_route: hop: sip:[EMAIL PROTECTED] set_destination: Parsing sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on for addr ess/port to
Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine
Hi I'm also interested in this. I currently have to press # to stop asterisk when I pick up the phone. Even if I answer before asterisk, it still answers. Cheers Rob On Sun, Dec 14, 2003 at 11:00:40PM -0500, Jim Flagg wrote: If Asterisk is configured as a simple answering machine replacement with the X100P connected to PSTN line. No FXS ports in the Asterisk machine. Standard phones are connect in parallel with the X100P like you would a regular answering machine. Can Asterisk detect that a phone has been picked up and cancel the outgoing message and/or voice recording? What about if the phones are connected to the pass-through port of the X100P? I know some PC software with voice modems can do this, just wondering if X100P/Asterisk can do it? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MGCP register
On Sat, 2003-12-20 at 03:22, Senad Jordanovic wrote: Hi, I am trying to figure out if * can register as a client on a remote MGCP service. Just like SIP and other protocols Do. Anyone tried this? No I don't believe it can. The MGCP implementation in Asterisk is a CallAgent not a UserAgent. --Karl Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Level(3) SIP termination services
John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! -dg -- Darnell Gadberry President binaryMedia darnell AT binmedia DOT com Date: Fri, 19 Dec 2003 21:12:22 -0500 To: [EMAIL PROTECTED] From: John Todd [EMAIL PROTECTED] Subject: [Asterisk-Users] Level(3) SIP termination services? Reply-To: [EMAIL PROTECTED] Anyone investigated the new service offerings from Level(3) in the last few months? They claim to be using ENUM and SIP - see http://www.level3.net/2192.html for details. Any idea of their pricing model for mid-sized enterprise applications or call centers for origination/termination? More specifically, do they interoperate with Asterisk? Some providers insist on certain hardware that speaks SIP flavor-of-the-month. I could call them to find out, but I suspect that this list will have far more clue than the Level(3) sales weasel that I'd get on the phone and who would want to waste a few days of my time asking stupid questions of me. Replies off-list, if you feel it necessary. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine
On Saturday 20 December 2003 02:13 pm, Robert Murray wrote: Hi I'm also interested in this. I currently have to press # to stop asterisk when I pick up the phone. Even if I answer before asterisk, it still answers. Cheers Rob Hello Rob. That's interesting. My system is/was like that. (I can still simulate it with the emergency phone that is paralleled/split at the incoming point). Pressing # certainly doesn't stop asterisk on my system. Asterisk CVS-12/10/03-11:49:50 Regards...Martin -- The world is coming to an end ... SAVE YOUR BUFFERS!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR sample config?
Rich Adamson wrote: Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes, I did look at the wiki and google searched for ivr menu.) Rich, I've started with the sample in the distribution. Check the file extensions.conf in the configs directory. And I'll add something to the wiki. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR sample config?
Joe Dennick wrote: Rich, This is in response to your question about and IVR Menu. Below is the dial-plan from the * at SAI. If you dial 1-800-747-9111 and the Extension 2998, you'll be able to hear this one in action. Now documented on http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+ivr+menu /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X101P + TDM400P
I thought I'd share my Asterisk experience, which hasn't exactly been as pleasant as I would like but now seems usable in most ways and more then I expected in other ways. I wanted a home PBX system, that would let me treat different callers different ways depending on CID. I initially bought the Digium developer's kit to try things out. That's a single port TDM400 and a X101P. I've added another X101P. One X101P terminates in a Vontage Cisco ATA-186. The other terminates with Qwest. The TDM400 is connected to both a TDD and a cordless phone. I also have a softphone connected along with 2 DID numbers through Voicepulse. I have a second Asterisk system outside my firewall to use for FWD. What went bad (most are minor): - SIP NATing. I just gave up on this. That's why I set up the second * box outside my firewall, with an IAX2 connection from my inside * box to it. - Voicemail - there are lots of little things missing. The main stuff is there, but lots of things I'd expect in a fully functional VM aren't there. I'm also disappointed that there is no way to turn on/off the MWI on a phone except through receiving a VM message (I'm working on a patch to add some basic MWI functionality for things like external VM systems) - TDM400 lockups - Sometimes, the TDM400 card seems to go into crazy static mode. It's the newest revision (this apparently is a known bug in some of the older versions). This hasn't happened since I moved cards around (see next item) and have updated the Asterisk software. The only solution was to unload and reload the wcfxs module. If it comes back, I suspect Digium will stand behind their product (I haven't contacted them formally, so if anything this is half my fault), and it may have been related to the IRQ problems. - IRQ/PCI problems. I have a lot of stuff in this machine that takes IRQs, including SCSI, sound card, net cards, etc. I have 6 filled slots right now. Initially, when I added the second X101P, the machine would not boot. It would either hang when the wcfxo module was loaded or crash with some weird SCSI IRQ errors. After Googling a big I found that some people can correct this by moving the cards around, so I did this. I also disabled serial and parallel ports in my BIOS. It took several tries at moving the cars around before I found a combination that works but things do seem to work right now even with the TDM400 card sharing an interrupt with the USB-UHCI device. - ECHO!!! I ended up updating the * code from CVS and turning on Mark2 with aggressive suppression. This fixed it, although things still sound a bit strange with full duplex talking (the echo suppression doesn't seem to like that and the voice volume changes along with some echo being present). I didn't have much of an echo problem with the X100 going into the ATA-186, but the X100 going to Qwest was miserable. Of course I also have DSL on the line, and the wiring in my house was done by the previous owner (who fancied himself as an electrician), so I'm not claiming my wiring is bad. Of course I never had any echo on my normal phones, nor did my DSL have any problems, so I do think there is something up with the X100 cards. Right now, echo seems okay. - Volume levels - I had to bump up the rxgain on the Qwest circuit a bit. But now all seems well. - Hangup and MWI clearing FSK tones - when hangup is executed after an extension dials out, * sends the FSK tones to clear the MWI (if appropriate). Unfortunately, while * may be hungup, I am not. WHAT WENT RIGHT: - IAX2 - this is slick. Works great between my inside-the-firewall * server and my outside-the-firewall * server, as well as between the inside box and Voicepulse. It would be nice if there was a *tad* better logging by default on incoming IAX calls (I had some problems initially with not having an entry for the Voicepulse DID lines, so * couldn't find the extension Voicepulse was looking for; rather then logging, it just hangs up; The IAX debug logs don't indicate *WHICH* extension is being looked for, either). - IVR functionality - that works great, too. No gripes at all - this is better then what you get with most PBX's - Unexpected functionality - I didn't know my cordless phone had a MWI. But it does. I was very surprised when a light I had never noticed on the phone before lit up after I received a voicemail message. That's a neat feature, and even neater that I didn't need to configure ANYTHING to get that to work. - Preliminary TDD support - this also pleased me, although there are some bugs in this support. It's nice to be able to set up a TDD interactive response system (right now, I can call my home machine, enter an IP address, and see my network management view of that machine through my TDD - which is also nice because lots of places have TDDs connected to pay phones...) [of course it would be nice if...* had a TDD extension that worked like the FAX extension if it heard TDD tones - especially
[Asterisk-Users] asterisk on beowulf cluster
Hi All, Can i install * on a beowulf cluster or Is * compatible to clusters. I am planning to install a 4 node beowulf cluster using few cheap hardwares. If no one had tried before i can spend some time on installing and configuring * on this cluster. Let me know. thanks, -Balaji Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing
Re: [Asterisk-Users] DIAX phone busy
Yes, IAX2 is checked. Thanks Dan wrote: Hi, Did you registered using IAX or IAX2? Check if in Registering page you have selected IAX2 or not. Did you check this one too? BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] x100P incoming
immediate=no did not work either, so at least to get * not write to CDR, I will have to use command noCDR(). My extension context is as follows [x100pincoming] exten = s,1,Ringing exten = s,2,Wait(40) exten = s,3,NoCDR() It still write the CDR in MySQL, any idea how to get rid of that ? SW -Original Message- From: SW [mailto:[EMAIL PROTECTED] Sent: Saturday, December 20, 2003 10:08 AM To: [EMAIL PROTECTED] Digium. Com Subject: Re: [Asterisk-Users] x100P incoming Exten = s,1,Wait(1000) This will make * not answering the call, but still you would see notices coming on your screen and also an entry in CDR. immediate=no Will try this out SW From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] x100P incoming Date: Sat, 20 Dec 2003 11:10:26 -0600 Reply-To: [EMAIL PROTECTED] On Friday 19 December 2003 08:12, David Gomillion wrote: How do I make x100P does not answer incoming calls ? The only thing that springs to mind is that you create an incoming context, and have an extension like: Exten = s,1,Wait(1000) Dunno if it will work or not, but that's the only thing that springs to mind. What about setting in zapata.conf: immediate=no before the channel declaration for the X100P. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk on beowulf cluster
On Sat, 2003-12-20 at 14:54, Balaji NJL wrote: Hi All, Can i install * on a beowulf cluster or Is * compatible to clusters. I am planning to install a 4 node beowulf cluster using few cheap hardwares. If no one had tried before i can spend some time on installing and configuring * on this cluster. Let me know. Just a bit of research would let you know that cluster can't migrate threads that use shared memory. Asterisk is such an app. So no asterisk wouldn't work on a cluster. A clever dialplan and some iax interconnects would make decent use of the hardware and you will learn more about the software. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX phone busy
Hi, Yes, IAX2 is checked. Then change the line to: exten = 70,1,Dial(IAX2/mike,30,tr) BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best SIP PHones to buy ?
- Original Message - From: Michael T Farnworth [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 21, 2003 4:31 AM Subject: Re: [Asterisk-Users] Best SIP PHones to buy ? We have bought around 30 Grandstream phones, both BT101 and BT102. In general the phone is reasonable, but it does have limitations. Notable issues for users tend to be the lack of any sort of consultative transfer or easy access to conference calling. Also its 'call waiting' facility is a rather annoying and loud normal ringing noise, rather than the usual 'beep beep' that people are probably used to and you can't disable the call waiting feature. Entering numbers also has problems as if you dial too quickly you tend to lose digits, even if you heard the tones and saw them appear on the display. Michael, I've put in a patch to fix call waiting for SIP phones, which is now part of the CVS. For each UA in sip.conf, insert the line incominglimit=1 Also, ensure that you have a usename=blah. This will stop the call waiting for the Grandstream phones. HTH. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine
On Sat, Dec 20, 2003 at 03:01:37PM -0500, Martin wrote: Hello Rob. That's interesting. My system is/was like that. (I can still simulate it with the emergency phone that is paralleled/split at the incoming point). Pressing # certainly doesn't stop asterisk on my system. It only works if you don't have a # extension, and no i extension. Another problem I have is that the x100p detects loud noise on the line as ringing. Cheers Rob Asterisk CVS-12/10/03-11:49:50 Regards...Martin -- The world is coming to an end ... SAVE YOUR BUFFERS!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine
That's interesting. My system is/was like that. (I can still simulate it with the emergency phone that is paralleled/split at the incoming point). Pressing # certainly doesn't stop asterisk on my system. It only works if you don't have a # extension, and no i extension. Another problem I have is that the x100p detects loud noise on the line as ringing. Use callprogress=no in the zapata.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] s, h, t, etc, extensions?
I'm in the process of reworking my dialplan to include an ivr and other items. I've seen several examples over the last several months that mention the s, h, t (and probably others) extensions, but I don't fully understand what they are used for. Can someone either give a short definition of each or point me towards some doc? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7912 speed dials
Hi ! We have Cisco 7912 phones, and the doc says that I can create up to four speed dial buttons on my phone using the Cisco CallManager. Does anyone knows which protocol is used to configure speed dials (Is it documented somewhere) ? Did someone tried to reverse engineer the protocol ? It would be cool, not having to pay $15000 just for configuring speed dials on those phones ;-D Cheers, -- Ludovic Drolez. http://www.palmopensource.com - The PalmOS Open Source Portal http://www.drolez.com - Personal site - Linux and PalmOS stuff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX phone busy
Yes, I've tried that as well. When I dial 70 from another extension, I hear ringing but the DIAX doesn't ring. Dan wrote: Hi, Yes, IAX2 is checked. Then change the line to: exten = 70,1,Dial(IAX2/mike,30,tr) BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine
What I mean, is that if I pick up the analogue phone connected in parallel with the x100p and blow into the handset, The x100p detects it as ringing and rings the phone connected to the s100u. Also, if I phone asterisk from another phone, and hang up when it is ringing, the s100u phone keeps ringing for several seconds. If I pick it up during this time, I get connected to the x100p and get a dial tone. Is it possible to make the x100p detect when the ringing stops quicker? Cheers Rob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine
On Saturday 20 December 2003 05:54 pm, Rich Adamson wrote: That's interesting. My system is/was like that. (I can still simulate it with the emergency phone that is paralleled/split at the incoming point). Pressing # certainly doesn't stop asterisk on my system. It only works if you don't have a # extension, and no i extension. Another problem I have is that the x100p detects loud noise on the line as ringing. Use callprogress=no in the zapata.conf Rich. That's unclear as you left my text as well as Roberts text. Are you talking about Another problem I have is that the x100p detects loud noise on the line as ringing. OR Pressing # certainly doesn't stop asterisk on my system. Regards...Martin -- It is bad luck to be superstitious. -- Andrew W. Mathis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] s, h, t, etc, extensions?
I'm in the process of reworking my dialplan to include an ivr and other items. I've seen several examples over the last several months that mention the s, h, t (and probably others) extensions, but I don't fully understand what they are used for. Can someone either give a short definition of each or point me towards some doc? s = start h = hangup t = timeout start is where the dialplan starts when no extension is given. hangup and timeout are pretty self-explanatory; they are where the dialplan jumps to on hangup or timeout, respectively. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] s, h, t, etc, extensions?
On Sat, 2003-12-20 at 16:57, Rich Adamson wrote: I'm in the process of reworking my dialplan to include an ivr and other items. I've seen several examples over the last several months that mention the s, h, t (and probably others) extensions, but I don't fully understand what they are used for. Can someone either give a short definition of each or point me towards some doc? s = Start. Used primarily for dialplans that enter a context with no other extension information. Think of a non DID phone line, call comes in, and we may only know that the line is ringing and nothing else. Even if you knew callerid, you have to still have a place to start. You can also think about s as a place to place part of the dialplan that you don't want callers to get back to unless they have passed through other functions. t = Timeout. Used for when calls have been inactive after a prompt was played. Also used to hang up a line that has been idle. h = Hangup. Used to clean up a call. Could be used to play a goodbye message before hanging up. Also seemingly used by the calling card people to record end of call for billing purposes. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] s, h, t, etc, extensions?
On a side note.. you can't use exten = h, if you have any hope of getting accurate billing info. Its wise to call ResetCDR(w) in your exten = h, or not use it at all. bkw On Sat, 20 Dec 2003, Steven Critchfield wrote: On Sat, 2003-12-20 at 16:57, Rich Adamson wrote: I'm in the process of reworking my dialplan to include an ivr and other items. I've seen several examples over the last several months that mention the s, h, t (and probably others) extensions, but I don't fully understand what they are used for. Can someone either give a short definition of each or point me towards some doc? s = Start. Used primarily for dialplans that enter a context with no other extension information. Think of a non DID phone line, call comes in, and we may only know that the line is ringing and nothing else. Even if you knew callerid, you have to still have a place to start. You can also think about s as a place to place part of the dialplan that you don't want callers to get back to unless they have passed through other functions. t = Timeout. Used for when calls have been inactive after a prompt was played. Also used to hang up a line that has been idle. h = Hangup. Used to clean up a call. Could be used to play a goodbye message before hanging up. Also seemingly used by the calling card people to record end of call for billing purposes. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZTMonitor - /dev/dsp problem
ztmonitor 1 -v On Sat, 20 Dec 2003, Daniel Bichara wrote: Hi, I am trying to run ZTMonitor to get debug info from my E100P board but I got the following message: -bash-2.05b# ./ztmonitor 1 Unable to open /dev/dsp: No such file or directory Cannot open audio ... -bash-2.05b# Thanks, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX phone busy
Yes,I often get the same result, but not always. - Original Message - From: Michael Welter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 20, 2003 3:40 PM Subject: Re: [Asterisk-Users] DIAX phone busy Yes, I've tried that as well. When I dial 70 from another extension, I hear ringing but the DIAX doesn't ring. Dan wrote: Hi, Yes, IAX2 is checked. Then change the line to: exten = 70,1,Dial(IAX2/mike,30,tr) BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine
Another problem I have is that the x100p detects loud noise on the line as ringing. Use callprogress=no in the zapata.conf That's unclear as you left my text as well as Roberts text. Are you talking about Another problem I have is that the x100p detects loud noise on the line as ringing. I had the loud noise problem with two x100p lines and someone suggested the callprogress=no parameter, which worked fine on both lines. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More beginner questions
--- Jon Creasey [EMAIL PROTECTED] wrote: Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Try this one the [fwd.pulver.com] section. Yes I see you have it in general. But fore some reason it needs to be there too disallow=all allow=ulaw allow=alaw Does anyone have any suggestions. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Level(3) SIP termination services
- Original Message - From: Darnell Gadberry [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 20, 2003 2:53 PM Subject: [Asterisk-Users] Level(3) SIP termination services John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! Maybe we can group buy :) My wife can use a hundred or so minutes... Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BYEXTENSION and DBPut
Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten = _*66XX,1,StripMSD,3 exten = _XX,2,DBPut,blacklist/BYEXTENSION/1 exten = _XX,3,Hangup However what I get in the database is: /blacklist/BYEXTENSION : 1 And BYEXTENSION is not replaced with the actual number dialed. Am I trying to do something that can not be done or am I just not doing it correctly? Walt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BYEXTENSION and DBPut
Walt Davis wrote:
Hey I need another pair of eyes on this!
I would like to add phones numbers to the blacklist from any handset so I
did this:
exten = _*66XX,1,StripMSD,3
exten = _XX,2,DBPut,blacklist/BYEXTENSION/1
exten = _XX,3,Hangup
However what I get in the database is:
/blacklist/BYEXTENSION : 1
And BYEXTENSION is not replaced with the actual number dialed.
Because BYEXTENSION is basically a Dial only feature.Use ${EXTEN}
Jeremy McNamara
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Re: [Asterisk-Users] BYEXTENSION and DBPut
Don't use BYEXTENSION use ${EXTEN}
bkw
On Sat, 20 Dec 2003, Walt Davis wrote:
Hey I need another pair of eyes on this!
I would like to add phones numbers to the blacklist from any handset so I
did this:
exten = _*66XX,1,StripMSD,3
exten = _XX,2,DBPut,blacklist/BYEXTENSION/1
exten = _XX,3,Hangup
However what I get in the database is:
/blacklist/BYEXTENSION : 1
And BYEXTENSION is not replaced with the actual number dialed.
Am I trying to do something that can not be done or am I just not doing it
correctly?
Walt
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[Asterisk-Users] ivr key press?
I'm testing an ivr implementation (first time) using: exten = 620,1,Wait,1 exten = 620,2,Answer exten = 620,3,DigitTimeout,5 exten = 620,4,ResponseTimeout,10 exten = 620,5,Background(npi-greeting) ; Thanks for calling press 1 for exten = 1,1,Goto(npi-directory,s,1) For initial testing, I've arbitrarily mapped this onto ext 620 (will change that later when things are working as expected). The initial npi-greeting message essentially says ...if you know your party's extension, you can dial it at any time. Press 1 for Sales, etc. If during this initial greeting I press 3000 (which is a valid extn), I can only press the first 3 before I get kicked out (I can't dial the full 3000). Am I supposed to be setting this up to expect only single-digit key presses (instead of 3000), or am I missing something that would suggest waiting for four key presses? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ivr key press?
On Sat, 2003-12-20 at 21:06, Rich Adamson wrote: I'm testing an ivr implementation (first time) using: exten = 620,1,Wait,1 exten = 620,2,Answer exten = 620,3,DigitTimeout,5 exten = 620,4,ResponseTimeout,10 exten = 620,5,Background(npi-greeting) ; Thanks for calling press 1 for exten = 1,1,Goto(npi-directory,s,1) For initial testing, I've arbitrarily mapped this onto ext 620 (will change that later when things are working as expected). The initial npi-greeting message essentially says ...if you know your party's extension, you can dial it at any time. Press 1 for Sales, etc. If during this initial greeting I press 3000 (which is a valid extn), I can only press the first 3 before I get kicked out (I can't dial the full 3000). Am I supposed to be setting this up to expect only single-digit key presses (instead of 3000), or am I missing something that would suggest waiting for four key presses? Well do you have a 3 extension in the npi-directory or any othe context being included? Posting more of your current extensions.conf file would help. Basically what you provided was much help as it isn't where you are in the dialplan when your problems occur, nor have you included the text from the console which will tell you where asterisk was trying to go. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] s, h, t, etc, extensions?
Care to expound a bit on that topic for the wiki, with some details as to why? JT At 6:32 PM -0600 12/20/03, Brian West wrote: On a side note.. you can't use exten = h, if you have any hope of getting accurate billing info. Its wise to call ResetCDR(w) in your exten = h, or not use it at all. bkw On Sat, 20 Dec 2003, Steven Critchfield wrote: On Sat, 2003-12-20 at 16:57, Rich Adamson wrote: I'm in the process of reworking my dialplan to include an ivr and other items. I've seen several examples over the last several months that mention the s, h, t (and probably others) extensions, but I don't fully understand what they are used for. Can someone either give a short definition of each or point me towards some doc? s = Start. Used primarily for dialplans that enter a context with no other extension information. Think of a non DID phone line, call comes in, and we may only know that the line is ringing and nothing else. Even if you knew callerid, you have to still have a place to start. You can also think about s as a place to place part of the dialplan that you don't want callers to get back to unless they have passed through other functions. t = Timeout. Used for when calls have been inactive after a prompt was played. Also used to hang up a line that has been idle. h = Hangup. Used to clean up a call. Could be used to play a goodbye message before hanging up. Also seemingly used by the calling card people to record end of call for billing purposes. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ivr key press?
Do you have the context with exten 3000 included in the same place as exten 620 is? If not then it can in no way work. bkw On Sat, 20 Dec 2003, Rich Adamson wrote: Hi Steve, On Sat, 2003-12-20 at 21:06, Rich Adamson wrote: I'm testing an ivr implementation (first time) using: exten = 620,1,Wait,1 exten = 620,2,Answer exten = 620,3,DigitTimeout,5 exten = 620,4,ResponseTimeout,10 exten = 620,5,Background(npi-greeting) ; Thanks for calling press 1 for exten = 1,1,Goto(npi-directory,s,1) For initial testing, I've arbitrarily mapped this onto ext 620 (will change that later when things are working as expected). The initial npi-greeting message essentially says ...if you know your party's extension, you can dial it at any time. Press 1 for Sales, etc. If during this initial greeting I press 3000 (which is a valid extn), I can only press the first 3 before I get kicked out (I can't dial the full 3000). Am I supposed to be setting this up to expect only single-digit key presses (instead of 3000), or am I missing something that would suggest waiting for four key presses? Well do you have a 3 extension in the npi-directory or any othe context being included? Posting more of your current extensions.conf file would help. Basically what you provided was much help as it isn't where you are in the dialplan when your problems occur, nor have you included the text from the console which will tell you where asterisk was trying to go. Okay, let me see if I can reword it a little different. While the exten = 620,5,Background(npi-greeting) statement is executing, I'm wanting to key in 3000 (as a valid extension). Should I be able to do that, or is the response limited to a single key press? If I attempt to do that, only the first digit is accepted. There are no CLI messages when the 3 is pressed. The C7960 drops the call. I don't care about dropping it right now as there is no submenu to handle a single key press of 3, therefore I'm expecting it to drop the call. The question is more oriented around is there some expectation within asterisk that the keypress is a single press, or should I expect to be able to enter the 3000 during the background message? If the answer is that I should be able to press 3000, then what might be causing this to immediately try to process the 3 and not wait for the full 3000? (That's that part I'm not seeing; forest and the trees kind of thing). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Level(3) SIP termination services
[crossposted to isp-clec and asterisk-users] Date: Fri, 19 Dec 2003 21:12:22 -0500 To: [EMAIL PROTECTED] From: John Todd [EMAIL PROTECTED] Subject: [Asterisk-Users] Level(3) SIP termination services? Reply-To: [EMAIL PROTECTED] Anyone investigated the new service offerings from Level(3) in the last few months? They claim to be using ENUM and SIP - see http://www.level3.net/2192.html for details. Any idea of their pricing model for mid-sized enterprise applications or call centers for origination/termination? More specifically, do they interoperate with Asterisk? Some providers insist on certain hardware that speaks SIP flavor-of-the-month. I could call them to find out, but I suspect that this list will have far more clue than the Level(3) sales humanoid that I'd get on the phone and who would want to waste a few days of my time asking stupid questions of me. Replies off-list, if you feel it necessary. JT At 11:53 AM -0800 12/20/03, Darnell Gadberry wrote: John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! -dg -- Darnell Gadberry President binaryMedia darnell AT binmedia DOT com That's not such a great price at 11 million minutes, in my opinion. Did you ask them if they would speak to Asterisk via SIP? We have on this list (asterisk-users) made some short lists of retail providers of minutes (see archives,) but it would be interesting to see what wholesale providers will warrant Asterisk use with their SIP gateways as acceptable. Questions for y'all to ask when you start fishing: - do you need dedicated interconnect with the network of the termination provider? In our example of L3, do you need to buy a fast ethernet of IP bandwidth from Level3, or will they take the traffic across a peer or other transit customer's link (in other words: the Internet.) - what codecs are supported? Some providers insist on G.711, which strikes me as underhanded at worst and significantly short-sighted, at best. If they support Asterisk (yay!) do they allow iLBC, GSM, and Speex? - how are CDR's transmitted back to the customer? Daily? Live? Monthly (agh!)? Via the Internet, or on tape/cd? - are media streams restricted to a single (or very few) IP addresses, or will they take media streams from anywhere? Again, many providers seem to want to break SIP's model of peer-to-peer media transmission, even when it's possible, for various business reasons. - are there any geographic considerations for sending traffic? Are there any area code/prefix sets which cost more? - are there any topological considerations for sending traffic? If traffic shifts between appearing on the East Coast, and then fails over (as an example) to coming in through a West Cost network perspective, does that change any SLAs or pricing? - if the provider offers multiple rates to North American destinations, how often does that rate table update? How is that rate table provided? - in what format does the provider offer international rate updates? How often? - does the provider offer private or public ENUM lookup for destinations locally served, if they provide DID's to their larger customer base? what is the provider's plan for ENUM rollout once NANP is fully ENUM capable? As you speak with providers, so long as there is no NDA, please share your experiences with the list. The medium-sized call-centric shops and call centers would benefit greatly from hearing the possible competitive alternatives that might be connected to Asterisk systems. This again brings up the topic of asterisk-biz mailing list, if there is any traffic. However, I doubt there will be much traffic since it seems the people who do many minutes keep their mouths shut, at least on this list. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFAX application
Hi sergio On Fri, 19 Dec 2003 14:49:15 +0100 Sergio Serrano Revuelto [EMAIL PROTECTED] wrote: Hi all, I have tested RxFAX application through X100P card. When Fax arrive i obtain the next trace: snip 5 (0.01679,-0.16590) - 0.02781 6 ( -0.04451, 0.75304) - 0.56904 7 ( -0.01415,-0.29305) - 0.08608 Fast carrier down Segmentation fault And i obtain 8 byte tif file. Any Idea? I have installed tiff-3.5.7 and spandsp-20031021. I get same result. but the end part looks like that. Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down -- Hungup 'Zap/1-1' with no segfault I'm tryed with tiff-v3.6.0 ( use with tar balled headers ) and spandsp-20031021 Does anyone have good result? Regards. mack_jpn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SJphone, Asterisk and DTMF tones ...
Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context = any) This is with dtmfmode=inband in sip.conf. With either rfc2833 or info, DTMF tones don't seem to get 'seen' by Asterisk at all. I'm running CVS-12/17/03-02:39:14, in case it's relevant. Help? -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 office +1.215.243.8335 fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Level(3) SIP termination services
Kudos John on an excellent set of questions! Not to mention the pointer to isp-clec. Thanks! John Todd wrote: [crossposted to isp-clec and asterisk-users] That's not such a great price at 11 million minutes, in my opinion. Did you ask them if they would speak to Asterisk via SIP? We have on this list (asterisk-users) made some short lists of retail providers of minutes (see archives,) but it would be interesting to see what wholesale providers will warrant Asterisk use with their SIP gateways as acceptable. Questions for y'all to ask when you start fishing: - do you need dedicated interconnect with the network of the termination provider? In our example of L3, do you need to buy a fast ethernet of IP bandwidth from Level3, or will they take the traffic across a peer or other transit customer's link (in other words: the Internet.) - what codecs are supported? Some providers insist on G.711, which strikes me as underhanded at worst and significantly short-sighted, at best. If they support Asterisk (yay!) do they allow iLBC, GSM, and Speex? - how are CDR's transmitted back to the customer? Daily? Live? Monthly (agh!)? Via the Internet, or on tape/cd? - are media streams restricted to a single (or very few) IP addresses, or will they take media streams from anywhere? Again, many providers seem to want to break SIP's model of peer-to-peer media transmission, even when it's possible, for various business reasons. - are there any geographic considerations for sending traffic? Are there any area code/prefix sets which cost more? - are there any topological considerations for sending traffic? If traffic shifts between appearing on the East Coast, and then fails over (as an example) to coming in through a West Cost network perspective, does that change any SLAs or pricing? - if the provider offers multiple rates to North American destinations, how often does that rate table update? How is that rate table provided? - in what format does the provider offer international rate updates? How often? - does the provider offer private or public ENUM lookup for destinations locally served, if they provide DID's to their larger customer base? what is the provider's plan for ENUM rollout once NANP is fully ENUM capable? As you speak with providers, so long as there is no NDA, please share your experiences with the list. The medium-sized call-centric shops and call centers would benefit greatly from hearing the possible competitive alternatives that might be connected to Asterisk systems. This again brings up the topic of asterisk-biz mailing list, if there is any traffic. However, I doubt there will be much traffic since it seems the people who do many minutes keep their mouths shut, at least on this list. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
On Sunday 21 December 2003 00:29, Darren Nickerson wrote: Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context = any) This is with dtmfmode=inband in sip.conf. With either rfc2833 or info, DTMF tones don't seem to get 'seen' by Asterisk at all. Changing the DTMF mode would indeed seem to be the logical solution. However, it appears that SJphone does not support that option (after a quick perusal of their PDF). You might want to file a bugtracker request on their website to implement that functionality. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
