[Asterisk-Users] Passing Argument to AGI
Can someone help me with passing argument to agi script. What I'm trying to this is execute agi script when hangup h,1,AGI(hangup.agi|string-argv-2348448) but can not get the argument variable passed to the hangup.agi script. I have tried $var = $ARGV[0]; or $var = $ARGV[1]; but still can not get the passing variable value. Thanks.
Re: [Asterisk-Users] Asterisk Codecs [G.729]
Rich, In real world, using real tool, getting real number. You don't expect to either talk only mode or listen only mode. Per call must have Rx Tx for inbound outbound. The numbers look more like 83Kbits/sec (thanks Andrew Gillham) in the wiki page or the cisco bandwidth consumsion (thanks Alex Volkow). I'm using IPTraf to monitor the bandwidth. When there is no call, it's 1.8 Kbits/sec. When making a call, it's 166.8 Kbits/sec ULAW codec: Total rates:166.8 kbits/sec 103.6 packets/sec Incoming rates: 83.6 kbits/sec 52.2 packets/sec Outgoing rates:83.2 kbits/sec 51.4 packets/sec Engineering rate is per channel but to calculate the bandwidth consumsion, it must be in realtime full-duplex. - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 10, 2004 3:41 AM Subject: Re: [Asterisk-Users] Asterisk Codecs [G.729] All of the numbers he's showing are apparently adding inbound and outbound traffic together, giving results that are approximately double what is actually seen on the wire. If he is working in a half-duplex ethernet environment, those numbers have some meaning; if full-duplex, then cut them in half for reasonable engineering values. (Also, some _appear_ to be questionable.) What is the method you are using to test the bandwidth. Can you give us a outline how to do a bit rate measurement on asterisk. snip ulaw 64 Kbps, sample-based Also known as alaw/ulaw 166kbits/sec alaw 64 Kbps, sample-based Also known as alaw/ulaw 167kbits/sec gsm 13 Kbps (full rate), 20ms frame size 66kbits/sec speex 2.15 to 44.2 Kbps n/a iLBC 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size 57.6kbits/sec G.729 8 Kbps, 10ms frame sizelicense Have anyone test it with G.729? Please let me know. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Codecs [G.729]
Absolutely agree, ITU standard is 64Kbits/sec. VoIP traffic with U-law per channel is 83Kbits/sec. VoIP traffic with U-law per call is 166Kbits/sec [measuring bandwidth per call] - Original Message - From: Nicolas Bougues [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 10, 2004 9:55 AM Subject: Re: [Asterisk-Users] Asterisk Codecs [G.729] On Wed, Mar 10, 2004 at 09:03:57AM -0800, Unavailable ID wrote: Rich, In real world, using real tool, getting real number. You don't expect to either talk only mode or listen only mode. Per call must have Rx Tx for inbound outbound. [...] Engineering rate is per channel but to calculate the bandwidth consumsion, it must be in realtime full-duplex. Sure, but in the real world, IP trafic is mostly carried over full duplex links (either serial or switched Ethernet), so you usually consider the trafic in one direction only (provided it's symetrical). If I consider an E1 to be 2 Mbps (2 inbound, 2 outbound, really), and Fast Ethernet to be 100 Mbps (again, 100+100 in full duplex), I shall consider U-law to be 83 kbits/s, not 166. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Codecs [G.729]
You're right. It's symmetric so it only takes 83Kbits/sec for u-law. IPTraf is confusing me :-) - Original Message - From: John Fraizer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 10, 2004 11:41 AM Subject: Re: [Asterisk-Users] Asterisk Codecs [G.729] You need to decide if you're going to measure both sides of the call or not. ITU standard is 64Kbits/s. That is correct. It is a standard DS0. But, guess what. That DS0 goes both directions so, measured bandwidth per call is 128Kbits/s using your logic. Only consumer grade DSL/Cable bandwidth is asymmetric. These wanna-be connections and the accompanying garbage that the sales/marketing monkeys spew (that the general public laps up as accurate) have done nothing but cause confusion to people who don't work the network side of the industry. [rent on] (Besides that, it makes their peering ratio so lopsided that there is no way they're going to get any decent no-settlement bilateral peering and as such, the price of those connections is going to remain artificially high since the providers have to PURCHASE transit instead of simply peering.) [rant off] When I order an OC3, it's 155Mb/s in BOTH DIRECTIONS. The industry doesn't say That's 310Mb/s of bandwidth because the measure of circuit bandwidth is by maximum one-way FLOW. You can (on paper) flow 155Mb/s in one direction on an OC3 (including encapsulation overhead). So, if you want to use accepted telco + IP metrics for measuring the flow (and thus the bandwidth), you look at the GREATER OF THE IN/OUT flow and that is how much bandwidth is required. ULAW = 83Kbits/s including encapsulation overhead. John wrote: Absolutely agree, ITU standard is 64Kbits/sec. VoIP traffic with U-law per channel is 83Kbits/sec. VoIP traffic with U-law per call is 166Kbits/sec [measuring bandwidth per call] - Original Message - From: Nicolas Bougues [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 10, 2004 9:55 AM Subject: Re: [Asterisk-Users] Asterisk Codecs [G.729] On Wed, Mar 10, 2004 at 09:03:57AM -0800, Unavailable ID wrote: Rich, In real world, using real tool, getting real number. You don't expect to either talk only mode or listen only mode. Per call must have Rx Tx for inbound outbound. [...] Engineering rate is per channel but to calculate the bandwidth consumsion, it must be in realtime full-duplex. Sure, but in the real world, IP trafic is mostly carried over full duplex links (either serial or switched Ethernet), so you usually consider the trafic in one direction only (provided it's symetrical). If I consider an E1 to be 2 Mbps (2 inbound, 2 outbound, really), and Fast Ethernet to be 100 Mbps (again, 100+100 in full duplex), I shall consider U-law to be 83 kbits/s, not 166. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Dial incomplete number
Hello all,
When I dial an international number, Asterisk dials
the incomplete number before I put in all digits.
For example, when I dial to 01135321XXX, it
pickups and dials 01135321 before I complete the dial.
If I press the redial, it will dial with all digits
that I put in before.
If I press all digits quick enough, it will dial
fine. Seems like * picks up to early.
-- Starting simple switch on
'Zap/46-1' -- Executing Dial("Zap/46-1",
"Zap/g1/01135321") in new stack -- Called
g1/01135321 -- Hungup 'Zap/2-1' == Spawn
extension (outbound, 01135321, 1) exited non-zero on
'Zap/46-1' -- Executing Hangup("Zap/46-1", "") in new
stack == Spawn extension (outbound, h, 1) exited non-zero on
'Zap/46-1' -- Hungup 'Zap/46-1'
What would I do to fix this?
Thanks.
Re: [Asterisk-Users] Asterisk Codecs [G.729]
It's full-duplex for both inbound outbound Total rates:166.8 kbits/sec 103.6 packets/sec Incoming rates: 83.6 kbits/sec 52.2 packets/sec Outgoing rates:83.2 kbits/sec 51.4 packets/sec - Original Message - From: Andrew Gillham [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, March 09, 2004 2:45 PM Subject: Re: [Asterisk-Users] Asterisk Codecs [G.729] Andrew Gillham wrote: Unavailable ID wrote: Hello all, I'm looking for advice for codec that works best for asterisk. Anyone has real testing with all codecs, specially with G.729. I have tested with single call on few codecs that come with asterisk by using IPTraf and the rate as of below: ulaw 64 Kbps, sample-based Also known as alaw/ulaw 166kbits/sec alaw 64 Kbps, sample-based Also known as alaw/ulaw 167kbits/sec gsm 13 Kbps (full rate), 20ms frame size 66kbits/sec speex 2.15 to 44.2 Kbps n/a iLBC 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size 57.6kbits/sec G.729 8 Kbps, 10ms frame sizelicense Have anyone test it with G.729? Please let me know. Thanks. Are some of these numbers for the full-duplex traffic? Ok, my question doesn't even seem that clear to me. :-) What I mean, is that the G.711 numbers for example look like both directions *combined* so the actual rate would be more like 83Kbit/s. (much like listed on the wiki page) So for G.711 a/ulaw, gsm, iLBC etc I was wondering if that is a single direction, or the combination of both directions? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Codecs [G.729]
Hello all, I'm looking for advice for codec that works best for asterisk. Anyone has real testing with all codecs, specially with G.729. I have tested with single callon few codecs that come with asterisk by using IPTraf and the rate as of below: ulaw64 Kbps, sample-based Also known as alaw/ulaw166kbits/secalaw64 Kbps, sample-based Also known as alaw/ulaw167kbits/secgsm13 Kbps (full rate), 20ms frame size66kbits/secspeex2.15 to 44.2 Kbpsn/aiLBC15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size57.6kbits/secG.7298 Kbps, 10ms frame sizelicense Have anyone test it with G.729? Please let me know. Thanks.
Re: [Asterisk-Users] G.729 vs. G.729 pass thru
Derek, Can you use fax with G.729? I know that only ULAW codec can use for fax but I don't know that if you can use fax with G.729 or not. BTW, what service provider that you are using? Quality can sometime depend on provider too. Thanks. - Original Message - From: Derek Samford [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 9:12 PM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru On all tests, I've run, fantastic. I haven't had any issues with voice quality at all, even on analog lines. Derek -Original Message- From: Unavailable ID [mailto:[EMAIL PROTECTED] Sent: Thursday, March 04, 2004 9:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru Darek, Thank you for the info. How is the sound quality when you are using with G.729 codec? What's your thought? Thanks. - Original Message - From: Derek Samford [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 2:58 PM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru As long as you have a IDE drive available, and mounted when you install it, it will work. This includes CD ROM's...It's what I did. Funkiness with the registration process. As far as pass through goes, from what I understand, it *should*, but when you have licensed binaries, from what I've seen, it doesn't. It's actually used 2 licenses. I plan on figuring that out next. Derek From: Unavailable ID [mailto:[EMAIL PROTECTED] Sent: Thursday, March 04, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G.729 vs. G.729 pass thru Hello everyone, If you don't have Digium card but you want to use G.729 codec, do you need a license for it? If the VoIP termination point supports G.729 and you are using sip phone (soft/hard phone), can you use the G.729 pass thru or you have to buy the license? Have anyone test it with SCSI system? Seems like it only work on machine with IDE disk. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 vs. G.729 pass thru
Hi Wes, Do you need to buy license when you are using pass thru. How does it work? I'm thinking about using pass thru for voip since the service provider has g.279 codec. Can you setup your * box connects to telco termination with pass thru? PBX =[T100P]= ASTERISK (*) =[G.729]= VOIP TERMINATION Thanks. - Original Message - From: Wes Marderness [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 05, 2004 11:42 AM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru I've had some small problems when trying to users features like AbsoluteTimeout with pass thru. Other than that sound quality has been good. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Unavailable ID Sent: Thursday, March 04, 2004 9:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru Darek, Thank you for the info. How is the sound quality when you are using with G.729 codec? What's your thought? Thanks. - Original Message - From: Derek Samford [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 2:58 PM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru As long as you have a IDE drive available, and mounted when you install it, it will work. This includes CD ROM's...It's what I did. Funkiness with the registration process. As far as pass through goes, from what I understand, it *should*, but when you have licensed binaries, from what I've seen, it doesn't. It's actually used 2 licenses. I plan on figuring that out next. Derek From: Unavailable ID [mailto:[EMAIL PROTECTED] Sent: Thursday, March 04, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G.729 vs. G.729 pass thru Hello everyone, If you don't have Digium card but you want to use G.729 codec, do you need a license for it? If the VoIP termination point supports G.729 and you are using sip phone (soft/hard phone), can you use the G.729 pass thru or you have to buy the license? Have anyone test it with SCSI system? Seems like it only work on machine with IDE disk. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Internet Phone Concept Question
Hi Greg, Welcome to * world :-) Your connection is slow '128k and upload speed of 32k' so you probably need the G.729 codec ($$$ - $10/channel/call from Digium). The X100P is only for dial-out from your phones that connect to TDM card. This should use to dial local number in Costa Rica. To call to US, use * to connect to the IAX service provider such as http://connect.voicepulse.com/ or http://www.nufone.net/ It looks like this: PSTN (USA) -- IAX Provider -- Asterisk -- TDM card -- Analog phones Hope this help. Tri Tu - Original Message - From: Greg Kedrovsky [EMAIL PROTECTED] To: asterisk-user [EMAIL PROTECTED] Sent: Friday, March 05, 2004 4:59 PM Subject: [Asterisk-Users] Internet Phone Concept Question Hey, all. I'm new to this Asterisk stuff and have a general concept question about making calls and whatnot over the net. I have a 4-port TDM card and a 1-port x100p card for incoming. All is configured and working fine. I have a _very_ simple configuration (start simple, add bells and whistles later). I have a cable modem hook-up and access the internet with a download speed of 128k and upload speed of 32k. My Asterisk server sits on a LAN, under a Freesco router running on an Pentium I machine (10BaseT cards because). I live in Costa Rica and would like to utilize the internet, if possible, to call family and friends in the U.S.A. Can I do that with Asterisk? Can I do that with standard analog phones through Asterisk? Can I do that without having another Asterisk machine State-side? If you have a link that would explain the concepts to me, that would be fine. Or if you could kinda prime the pump for me so I can get the ball rolling on my end - that'd be very much appreciated, too. Thanks ahead of time. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound fax using T100P
Hello everyone, Is there anyone know how to make fax detection work with the T100P either to regular PSTN or VOIP? I'm having problem with sending fax out. I have two T100P cards which connects to T1 PSTN and the other connects to PBX (T100P -- Asterisk [T100P] -- PBX). Thanks. -Tri.
[Asterisk-Users] Outbound fax with T100P
Hi Mark,
I'm having problem with fax detection on my Asterisk
box. My config is like this:
PSTN T1 == Asterisk (*) T100P (2 cards) ==
PBX
Everything works fine for voice and incoming fax but out
going fax got this error:
-- Starting simple switch on
'Zap/48-1'
-- Executing
Dial("Zap/48-1", "IAX2/myusername@NuFone/16509300206") in new
stack
-- Called myusername@NuFone/16509300206
-- Call accepted by 66.225.202.72 (format
ULAW)
-- Format for call is
ULAW
-- IAX2[NuFone]/2 stopped
sounds
-- Redirecting Zap/48-1 to fax
extension
Mar 4 12:11:58 WARNING[1217602880]: app_dial.c:293
wait_for_answer: Unable to forward
frame
-- Hungup
'IAX2[NuFone]/2'
== Spawn extension (outbound, fax, 0) exited non-zero on
'Zap/48-1'
-- Executing Dial("Zap/48-1", "Zap/g1/fax") in new stack
-- Called g1/fax
-- Hungup
'Zap/1-1'
== Spawn extension (outbound, fax, 1) exited non-zero on
'Zap/48-1'
-- Executing
Hangup("Zap/48-1", "") in new stack
== Spawn extension (outbound, h, 1) exited non-zero on
'Zap/48-1'
-- Hungup
'Zap/48-1'
Is there
something wrong here?
[outbound]
; Fax extention
Dial Plan;exten = fax,1,Dial,${PSTN}/${EXTEN}
[default]
include =
inboundinclude = outbound
Thanks.
Tri
Tu
[Asterisk-Users] G.729 vs. G.729 pass thru
Hello everyone, If you don't have Digium card but you want to use G.729 codec, do you need a license for it? If the VoIP termination point supports G.729 and you are using sip phone (soft/hard phone), can you use the G.729 pass thru or you have to buy the license? Have anyone test it with SCSI system? Seems like it only work on machine with IDE disk. Thanks.
Re: [Asterisk-Users] G.729 vs. G.729 pass thru
Darek, Thank you for the info. How is the sound quality when you are using with G.729 codec? What's your thought? Thanks. - Original Message - From: Derek Samford [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 2:58 PM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru As long as you have a IDE drive available, and mounted when you install it, it will work. This includes CD ROM's...It's what I did. Funkiness with the registration process. As far as pass through goes, from what I understand, it *should*, but when you have licensed binaries, from what I've seen, it doesn't. It's actually used 2 licenses. I plan on figuring that out next. Derek From: Unavailable ID [mailto:[EMAIL PROTECTED] Sent: Thursday, March 04, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G.729 vs. G.729 pass thru Hello everyone, If you don't have Digium card but you want to use G.729 codec, do you need a license for it? If the VoIP termination point supports G.729 and you are using sip phone (soft/hard phone), can you use the G.729 pass thru or you have to buy the license? Have anyone test it with SCSI system? Seems like it only work on machine with IDE disk. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] wct1xxp module and the T100P
Use default configuration as below should work if you have PRIline. zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us zapata.conf: [channels] context=default switchtype=national signalling=pri_cpe channel=1-23 group = 1 If it's not green, make sure you put in right framing, coding, signalling, switchtype. Call your telco and ask them if you do not know for sure. It could be EM Wink (channel bank line) which uses different settings. Hope this help. Tools: /sbin/ztcfg or /sbin/zttool -Tri. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 03, 2004 3:28 PM Subject: Re: [Asterisk-Users] wct1xxp module and the T100P Andrew McRory [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 03/03/2004 04:11 PM Please respond to asterisk-users To: [EMAIL PROTECTED] cc: Subject:Re: [Asterisk-Users] wct1xxp module and the T100POn Wed, 3 Mar 2004 [EMAIL PROTECTED] wrote: I'm having trouble turning up a PRI to a T100P. I've read on the Digium FAQ's that once the wct1xxp module is loaded correctly, the LED on the T100P will flash red. I believe I've loaded the module correctly because both wct1xxp and zaptel are listed when I do the "lsmod" command. The LED on the card does not flash on and off. Does anyone have any recommendations on what I could be doing wrong?Switch type, line code, framing all matter. How about posting your config?-- Andrew McRory - President/CTOLinux Systems Engineers, Inc.PO BOX 3791Tallahassee, FL 32315(850)224-5737(850)294-7567___ Sure: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us zapata.conf: [channels] context=cyperpri switchtype=national pridialplan=unknown signalling=pri_cpe channel=1-23 Here's a question, though: does the wct1xxp module read from either zaptel.conf or zapata.conf when loaded? Thanks! chris
[Asterisk-Users] Block Callerid with VoicPulse Connect!
Hi everyone, Is there anyone know how to block callerid with VoicePulse Connect outbound termination? It's showing up 000-000- when I call out. Please show me the trick if you know how. Thanks. -Tri.
Re: [Asterisk-Users] Block Callerid with VoicPulse Connect!
Thanks Andrew. But it doesn't work either. I believe that this must be done at the termination end point (VoicePulse Asterisk Server) since the outbound is going to VoicePulse PSTN line (PRI/T1). They must set that option on their end. However, technical support from VoicePulse said that they are working on it to make it possible in the future. What's a pain... -Tri. - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 01, 2004 11:52 AM Subject: RE: [Asterisk-Users] Block Callerid with VoicPulse Connect! Unavailable ID wrote: Hi everyone, Is there anyone know how to block callerid with VoicePulse Connect outbound termination? It's showing up 000-000- when I call out. Please show me the trick if you know how. Thanks. -Tri. If you'll do a SetCallerID(your_number_here) you can set it to any number you wish. I read something a day or so about setting the restricted flag when calling over a PRI, but I've just been catching up on all the * mail so there's no telling when the email actually was sent. Anyway, they said set this as your callerid line for the device in (sip,h323,iax,or whatever they were working with) Callerid=Anonymous 910997 ; number used for clarity purposes ; don't use this number, I made it up! - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Block Callerid with VoicPulse Connect!
Hi Matt, Thank you for the info. Have you test with the SetCallerID(your_number_here) to something likes SetCallerID(Unavailable ID) with your callerid? I don't actually want to show my caller ID. Please let me know if you can do that or there is an option to block caller ID from their website after you purchased a DID. Thanks. -Tri. - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 01, 2004 3:19 PM Subject: Re: [Asterisk-Users] Block Callerid with VoicPulse Connect! We had the same problem... we were trying to set caller id to 6434701641 (64 for NZ 3 for Dunedin rest for our number) it didn't work... However, we purchased a DID line for voicepulse and set the cid to that and it worked. Maybe because the number was a US number and owned by voicepulse? Matt - Original Message - From: Unavailable ID [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, March 02, 2004 9:57 AM Subject: Re: [Asterisk-Users] Block Callerid with VoicPulse Connect! | Thanks Andrew. But it doesn't work either. | | I believe that this must be done at the termination end point (VoicePulse | Asterisk Server) since the outbound is going to VoicePulse PSTN line | (PRI/T1). They must set that option on their end. | | However, technical support from VoicePulse said that they are working on it | to make it possible in the future. | | What's a pain... | | -Tri. | | - Original Message - | From: Andrew Thompson [EMAIL PROTECTED] | To: [EMAIL PROTECTED] | Sent: Monday, March 01, 2004 11:52 AM | Subject: RE: [Asterisk-Users] Block Callerid with VoicPulse Connect! | | | Unavailable ID wrote: | Hi everyone, | | Is there anyone know how to block callerid with VoicePulse Connect | outbound termination? It's showing up 000-000- when I call out. | Please show me the trick if you know how. | | Thanks. | | -Tri. | | If you'll do a SetCallerID(your_number_here) you can set it to any | number you wish. | | I read something a day or so about setting the restricted flag when | calling over a PRI, but I've just been catching up on all the * mail so | there's no telling when the email actually was sent. | | Anyway, they said set this as your callerid line for the device in | (sip,h323,iax,or whatever they were working with) | | Callerid=Anonymous 910997 ; number used for clarity purposes | ; don't use this number, I made it up! | | - | Andrew Thompson | http://aktzero.com/ | | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Block Callerid with VoicPulse Connect!
Matt, This is what I'm looking for. Our current PBX system is a supertrunk (channel bank) that doesn't have Caller ID. When we switched to VoIP, it shows 000-000- and we don't want that. So adding a DID solve this issue. I will go and buy it. Thank you very much. -Tri. - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 01, 2004 3:54 PM Subject: Re: [Asterisk-Users] Block Callerid with VoicPulse Connect! :-) The company I'm doing that for is a telemarketing company. By law they have to send caller id with every call. Before we did this, there was no caller id sent (i.e. it was blocked) Soto block it you have to do nothing... Matt - Original Message - From: Unavailable ID [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, March 02, 2004 12:49 PM Subject: Re: [Asterisk-Users] Block Callerid with VoicPulse Connect! | Hi Matt, | | Thank you for the info. | | Have you test with the SetCallerID(your_number_here) to something likes | SetCallerID(Unavailable ID) with your callerid? I don't actually want to | show my caller ID. Please let me know if you can do that or there is an | option to block caller ID from their website after you purchased a DID. | | Thanks. | | -Tri. | | - Original Message - | From: Matt Riddell [EMAIL PROTECTED] | To: [EMAIL PROTECTED] | Sent: Monday, March 01, 2004 3:19 PM | Subject: Re: [Asterisk-Users] Block Callerid with VoicPulse Connect! | | | We had the same problem... | | we were trying to set caller id to 6434701641 | (64 for NZ 3 for Dunedin rest for our number) | it didn't work... | | However, we purchased a DID line for voicepulse and set the cid to that | and it | worked. | | Maybe because the number was a US number and owned by voicepulse? | | Matt | - Original Message - | From: Unavailable ID [EMAIL PROTECTED] | To: [EMAIL PROTECTED] | Sent: Tuesday, March 02, 2004 9:57 AM | Subject: Re: [Asterisk-Users] Block Callerid with VoicPulse Connect! | | | | Thanks Andrew. But it doesn't work either. | | | | I believe that this must be done at the termination end point | (VoicePulse | | Asterisk Server) since the outbound is going to VoicePulse PSTN line | | (PRI/T1). They must set that option on their end. | | | | However, technical support from VoicePulse said that they are working on | it | | to make it possible in the future. | | | | What's a pain... | | | | -Tri. | | | | - Original Message - | | From: Andrew Thompson [EMAIL PROTECTED] | | To: [EMAIL PROTECTED] | | Sent: Monday, March 01, 2004 11:52 AM | | Subject: RE: [Asterisk-Users] Block Callerid with VoicPulse Connect! | | | | | | Unavailable ID wrote: | | Hi everyone, | | | | Is there anyone know how to block callerid with VoicePulse Connect | | outbound termination? It's showing up 000-000- when I call out. | | Please show me the trick if you know how. | | | | Thanks. | | | | -Tri. | | | | If you'll do a SetCallerID(your_number_here) you can set it to any | | number you wish. | | | | I read something a day or so about setting the restricted flag when | | calling over a PRI, but I've just been catching up on all the * mail | so | | there's no telling when the email actually was sent. | | | | Anyway, they said set this as your callerid line for the device in | | (sip,h323,iax,or whatever they were working with) | | | | Callerid=Anonymous 910997 ; number used for clarity purposes | | ; don't use this number, I made it up! | | | | - | | Andrew Thompson | | http://aktzero.com/ | | | | | | ___ | | Asterisk-Users mailing list | | [EMAIL PROTECTED] | | http://lists.digium.com/mailman/listinfo/asterisk-users | | To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | | | ___ | | Asterisk-Users mailing list | | [EMAIL PROTECTED] | | http://lists.digium.com/mailman/listinfo/asterisk-users | | To UNSUBSCRIBE or update options visit: | |http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL
