Re: [Asterisk-Users] Codec Conversion
Doesn't g729 require a license? Lyle - Original Message - From: Sean Cook [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 02, 2004 8:12 PM Subject: Re: [Asterisk-Users] Codec Conversion I think that all you have to do is where you define the codecs for the extention/protocol and asterisk will take care of the rest... [sip2101] [sip2102] allow=g711allow=g729 Asterisk will make the conversion on its own... I could be wrong but I think that is the way it works Sean kido noagbodji wrote: Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is there is a general way of doing that. Thanks K. ___ 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] Broadvoice outbound 404 error
1. Have you contacted Broadvoice Technical Support before sending mail in this list? 2. Broadvoice uses this list for some reason to market their products creating situations like this. Is there no one to control on this list. If you don't get proper technical support, you have to decide what to do? -Kannaiyan - Original Message - From: Reid Forrest [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, December 04, 2004 10:22 PM Subject: [Asterisk-Users] Broadvoice outbound 404 error Is anyone else experiencing 404 errors on outbound dial with Broadvoice? I've followed the instructions posted by Broadvoice to configure sip.conf, and inbound calling works fine. Every time I try to dial out, I get a 404 Not Found error. Here are the relevant sections from my configs. sip.conf: context=broadvoice-in register = [EMAIL PROTECTED]:xxpasswordxx:[EMAIL PROTECTED] [bv-home] type=peer host=proxy.dca.broadvoice.com fromdomain=sip.broadvoice.com fromuser=3215551212 context=inbound canreinvite=no qualify=yes disallow=all allow=ilbc allow=gsm allow=ulaw dtmfmode=inband secret=xxpasswordxx insecure=very Thank you, Reid Forrest, CISSP Max-IS, Inc. [EMAIL PROTECTED] ofc: 407.786.9600 x1200 cell: 321.439.8903 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] x100p offhook/onhook states
Hi, I'm having an interesting problem with my card. It seems to work fine, for the most part. When I first load the module and asterisk, it detects the line in the on-hook state. However, after the first phone call, zap show channel 1 lists it as being off-hook. During subsequent calls, the card is listed as being on-hook, and when it's not used -- off-hook. There are also some weird problems with detection when the remote side hangs up -- sometimes it works, other times it doesn't. However the times when it doesn't are usually when the other side is a cell-phone. (Does this matter?) Does this off/on-hook reversal stuff matter? Is there any reason that remote disconnect would not work *some* of the time? I checked the line, and the voltage drops when the remote side hangs up (checked with a regular phone with an LED-lit keypad). What settings should I be playing around with? Is it OK if other phone devices are hooked up to the same line if they are never used? Does a phone plugged in actually change any characteristics of the line if it's never picked up? The card is actually a clone card, and though I realise that a lot of people on this list are strongly against the people who use clones, I urge you to consider this question as if it were an actual digium x100p card, and not just write it off as a clone-related problem. Of course, if there are no good explanations, I'm willing to accept that as the cause. Thanks, Ilia Mirkin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FOP Asterisk Manager Login Failed?
On Fri, 03 Dec 2004 18:49:32 -0500, Nick Bachmann [EMAIL PROTECTED] wrote: Noah Miller wrote: I've told lots of people about the Flash Operator Panel before, but I've never actually used it myself. I've got it all set up nicely, but I can't seem to authenticate to the asterisk manager (which is running on the same box). When I set the op_server.pl to give debug messages, it shows that it can reach the asterisk manager, but cannot authenticate: ** Asterisk event received, process block... - Action: Login - Username: user - AuthType: MD5 - Key: 0be2f6f6e39f05a53f5a292517ede3e2 ** End of block - Response: Error - Message: Authentication failed I note that it says the authentication is done with MD5, do I need to put an MD5 hash in for the secret in the configuration files? No. The md5 is used so that your actual secret does not have to be transmitted in plaintext. The concatination of the random key and the secret is computed by both sides and hashed, if these two intermediate forms of your secret are the same you are authenticated. [user] secret = usersecret deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 Is your FOP on a different machine? If so, you'll have to explicitly add its IP or remove the deny statement, as it is blocking all IPs on all subnets. If it is on the same machine also include the IP Address as well as 127.0.0.1 Jason ___ 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] Snom 220 busy lamps [was: Receptionist phone...]
On Sat, Dec 04, 2004 at 11:51:24AM +0100, Peter Svensson spake thusly: I guess it may just be a typo during retyping, but you have 'l' (lower case L) in the hint line and a '1' (one) in the macro line. SON OF A [EMAIL PROTECTED]@[EMAIL PROTECTED]@#^*$#%@@ ahem You are correct, somehow when I put in the hint lines the 1 became an l so every single one of my hint extensions is incorrect. I am pretty sure I cut and pasted this from the wiki or someones email so I am looking for a possible error there. I have no idea how that happened as the 1 and l keys are nowhere near each other on the keyboard although they look very nearly identical in Linux terminal font so I never would have caught it. One thing that still concerns me is that when I reload the dialplan it still says the priority is -1 : -- Added extension '100' priority -1 to default when it seems like it should say priority hint. I logged in remotely to fix the dialplan but won't know for sure if the busy lamps are working properly until Monday when I can physically get into the office. Thanks for the extra pair of eyes! -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpSai6IEZT6V.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receptionist Phone
On Sat, Dec 04, 2004 at 08:03:03PM -0400, Cian O'Sullivan spake thusly: She is an older lady and does not want to use a web interface. Any suggestions? Give her a Snom or Polycom phone which does have this capability and set it up like this: http://lists.digium.com/pipermail/asterisk-dev/2004-August/005917.html After all of the debating on this list over receptionist phone it turns out that * can do it afterall, you just need a decent phone. I just implemented it myself remotely although I'll have to actually go into the office on Monday to see if the buttons are really working. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpAtGXgVDMQJ.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receptionist Phone
Cian O'Sullivan wrote: I have a customer interested in an * system, however she wants to ensure that the receptionist phone will display who is on the phone and who is not. It is an office of 10 people, and there are 12 PRI channels available. She is an older lady and does not want to use a web interface. Any suggestions? Using a Cisco with a XML browser and a CGI generated image, of who is on the phone at that time. Probably enough space to fit 10 persons in with a shrunk down font. Stefan de Konink ___ 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] Is Gigabit Ethernet necessary?
[EMAIL PROTECTED] wrote: For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100 Mbps should be fine... The office will have 20 phones, with remote VoIP phones added to the mix later on. http://www.voip-calculator.com/calculator/lipb/ Don't forget that you can't send 100 Mbps through a 100Mbps link. TIA, TR41, probably. Nick ___ 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] BLOCKING incoming FAXES on voice line.
At time to time somebody is trying their luck and send me most likely a junk fax on my voice line. During normal working hours is not a problem I just pickup the line and hangup the call but after-hours my voice mailbox is intercepting the call and recording those beeps (waisting my CPU cycles). Is there a way to block call / issue hangup command if the incoming call is a fax? -- #Joseph ___ 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] Two zaptel T1 cards: no clock from one
Yeah, proper crossover cable. I've eliminated all cabling issues with the T1 analyzer. I get a full and accurate pattern back when I test from the cable end where it would have been connected into the T100P, with the channel bank in loopback. The main symptom is that when I hook the analyzer directly to either the channel bank or the T100P, neither is providing clock. I could have the channel bank supply one, but I will have fax/modem calls bridged between the two PCI cards, so a common clock is best. The most disturbing thing is that the T100P, as the only card in a system, provided clock just fine. There was a thread last month in -dev about being unable to use common clock source across cards. Is this related? How can one cause zaptel to provide ref clock? Should I be seeing 1000 interrupts/sec on any and all TDM cards? Help me understand what you mean by neither is providing clock. By definition, every single T1 provides clocking within the transmit side of a T1. Its embedded in the data stream and you can't turn it off. Are you talking about clock sync? ___ 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] Polycom 500, won't ring??
Um, do you mean ipmid.cfg and sip.cfg? Did you follow these instructions?: http://www.voip-info.org/wiki-Polycom+auto-answer+config John Peter Johnson wrote: You might want to check your phone directory file. In there you can specify a ring type for a identified incoming caller - perhaps you have specified ring type 0 which is by default silent. Peter -Original Message- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Jared Armstrong *Sent:* Saturday, 4 December 2004 8:31 AM *To:* [EMAIL PROTECTED] *Subject:* [Asterisk-Users] Polycom 500, won't ring?? Hi, I have was testing some of the different ring types with my polycom 500, and the ALERT_INFO settings. Now when my phone receives a call it wont ring. All the other phones ring fine, and my phone wasnt the only one I rebooted with the changed sip.conf and impd.conf. I have reverted back to a standard sip.conf and impd.conf and I still can not get my phone to ring for any incoming calls. Does anyone have any suggestions to look for? Jared Armstrong ___ 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] Billing - which program are you using?
I want to play around with post billing. List of all phone calls, ... Which program is useful for that? All what I have seen are not based on CDR, but on Radius. What are you using? bye Ronald ___ 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] Help with music over intercom.
Christopher Dobbs wrote:
I am using Console/DSP for an intercom. I want to play my MP3
collection over it when no one is using it, like when they do in the
supermarket.
I doubt this code will work if you cut and paste, as I'm just writing
from memory to give you an idea of what I would do. First create your
intercom extension.
exten = 555,1,Hangup(Console/DSP)
exten = 555,2,Dial(Console/DSP,,g)
exten = 555,3,System('script to copy moh.call to moh2.call, then move
it to the spool/outgoing folder')
the g in Dial is to go on when the caller hangs up -- you will want to
double check g is in fact the right flag.
Your moh.call file should be written to call Console/DSP and connect it
to the MusicOnHold application.
Then the MOH will stop when extension is called, so you can make an
announcement, and will resume when the announcer hangs up when another
call is placed to the MOH generator.
Good luck,
Trevor Peirce
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Re: [Asterisk-Users] asterisk dabbling...
Ray Jender wrote: Newbee here I would like to play around with Asterisk a little. First, I need to prepare a server with FreeBSD. It's a PII 433mHz/256mb box. Good enough? Then install Asterisk. I have a broadband (cable) internet presence. Could I do anything with this connection and Asterisk? Thanks, Rayasterisk Ray, I hate to say this (I am a huge FreeBSD fan), but I believe that each OS has it's own strengths. While FreeBSD isn't any better or worse than Linux for Asterisk, Linux was the platform that it was originally developed on. It sounds like you are new so I will suggest that you stick with Linux for now and enjoy more support options, better hardware support, and more documentation. I have run Asterisk on both (even inside a FreeBSD jail) and I will say that I prefer to run it in Linux because as of now it just works better. My web servers, mail servers, etc, etc, etc. can run FreeBSD because I happen to like FreeBSD for those tasks. But not for running * (as of now, that could change...) That hardware should be fine, but then again I don't even know what you will be doing with it. There are people that run * on P133's. But like anything else, don't expect it to be able to work magic just because it is Linux and OSS. Hardware limits are still hardware limits. I would say though, that for most of the common stuff that you will want to play around (dabble) with, this machine sounds fine (some would say more than fine). I have run it on much less http://www.krisk.org/astlinux/ As for cable internet, it all depends. How much bandwidth do you have, are you behind NAT? What kind of packet loss/latency/jitter do you typically experience? If I were you I would just give it a shot and see how it works! P.S. - use kernel 2.6 if you can -- Kristian Kielhofner ___ 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 Hardware
Can I start using Asterisk with a couple of SIP IP phones and Softphone software on users PCs only? I do not have any cards yet and will still have to wait until I order a card. Regards,Walid ___ 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] Two zaptel T1 cards: no clock from one
On Sat, 4 Dec 2004 21:18:50 -0600, Rich Adamson [email protected] wrote: Help me understand what you mean by neither is providing clock. By definition, every single T1 provides clocking within the transmit side of a T1. Its embedded in the data stream and you can't turn it off. Are you talking about clock sync? First, In reply to Lyle, framing/line coding match, and the span 5 is in the dmesg, but the wct1xxp doesn't spit out SPAN x and it's mixed in: Found a Wildcard: Digium Wildcard T100P T1/PRI ...Using ESF/B8ZS coding/framing You're absolutely right, so I must be thinking of clock sync, and I'm not clear on recall what I saw from the channel bank. From the T100P though I am sure I saw zero hertz rx. I'll verify again Monday. Anyway, I'll be wrangling Digium again Monday (support was unresponsive Friday). Regardless of other issues, I expect it should be clearing alarms when I stick a loopback plug in. ztcfg (snipped a bit): pbx:~# ztcfg -vv Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 5: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: (snipped out 24,48,72 = D, 1-23,25-47,49-71, 73-94 = B) Channel 95: Individual Clear channel (Default) (Slaves: 95) Channel 96: D-channel (Default) (Slaves: 96) Channel 97: FXO Kewlstart (Default) (Slaves: 97) Channel 98: FXO Kewlstart (Default) (Slaves: 98) Channel 99: FXO Kewlstart (Default) (Slaves: 99) Channel 100: FXO Kewlstart (Default) (Slaves: 100) Channel 101: FXO Kewlstart (Default) (Slaves: 101) Channel 102: FXO Kewlstart (Default) (Slaves: 102) Channel 103: FXO Kewlstart (Default) (Slaves: 103) Channel 104: FXO Kewlstart (Default) (Slaves: 104) Channel 105: FXO Kewlstart (Default) (Slaves: 105) Channel 106: FXO Kewlstart (Default) (Slaves: 106) Channel 107: FXO Kewlstart (Default) (Slaves: 107) Channel 108: FXO Kewlstart (Default) (Slaves: 108) 108 channels configured. zttool (sucky paste): x OK TE410P (PCI) Card 0 Span 1 a x x OK TE410P (PCI) Card 0 Span 2 # x x OK TE410P (PCI) Card 0 Span 3 a x x OK TE410P (PCI) Card 0 Span 4 a x x RED Digium Wildcard T100P T1/PRI Card 0 a x x xCurrent Alarms: Red Alarm x x x xSync Source:Internally clocked x a x x xIRQ Misses: 0 x a x x xBipolar Viol: 0 x a x x xTx/Rx Levels: 0/ 0 x a x x xTotal/Conf/Act: 24/ 12/ 0 x a x x x 112lqqkx a x x x123456789012345678901234x Back xx a x x xTxA mqqjx a x x xTxB x a x x xTxC x # x x xTxD x x x xlqqkx x x xRxA x Loop xx x x xRxB mqqjx x x xRxC x x x xRxD x ___ 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] Sveasoft Alchemy QOS
On Wed, 1 Dec 2004, Kanuri, Seshu (Company IT) wrote: Tell me which one can get me access to the LinkSys Linux using SSH? Does Satori has this feature? I am not so concerned with Voice Shaping and QOS at this time, but more interested in converting this into a Linux box that is accessible from an ssh client. Alchemy has ssh access, you need to pay $20 subscription to Sveasoft to access the pre-release firmware. Steve - $20.00 for GNUed hackware that is originally freely donated by LinkSys? No way. Well, then roll your own and stop whining about it. Quite frankly calling it hackware shows that you have no concept of how much work has gone into the Sveasoft firmware, nor do you grasp the concept that Linksys is incorporating many of the Sveasoft changes BACK into their firmware. Everyone wins from this, and Sveasoft has a revenue stream that allows them to keep focused development on improving the firmware. I have over 60 of the WRT54GS units in production and I run Sveasoft firmware on every single one of them. It is so far ahead of Linksys's internal builds and adds so many additional features that there is no comparison between the two. Hackware indeed. What an insult to all of the quality developers that are putting their time and effort into extending the platform and making it one of the most incredible sub $100 routing platforms on the planet. How much did I pay for Asterisk? Was it $20 grand? Don't remember having paid that much for Asterisk. $20 US DOLLARS! Not, $20,000! If not SSH what other way can we access Linksys Linux without Alchemy wrapper sftp/rlogin/telnet? Does anyone know? Plenty of people know. If you can't do a google search for it, this isn't the place to be asking about it. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Is Gigabit Ethernet necessary?
[EMAIL PROTECTED] wrote: For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100 Mbps should be fine... The office will have 20 phones, with remote VoIP phones added to the mix later on. TIA, -Ron Ron, For what it costs, it is usually worth it to put a gig card in your server (a good one). Gigabit cards have newer and much better buffering and pci bus support. They are also much better at offloading processing from the system's CPU. You need to make sure that you have a good one. Because a crappy Gigabit card is probably not much better than a crappy 100mb card... I like Intel nics (both 100mb and 1000mb). Something supported by e1000 shouldn't be too expensive and usually will work pretty well with most all OS's. my $0.02 -- Kristian Kielhofner ___ 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] ZAP and IAX Trunks
Thanks Dean.. Well, about the hardware then. What do you recommend for beginning with Asterisk. I intend to use Cisco 7940s/7960s with Asterisk. Also which software is recommended to enable Soft phone on users PCs? Regards, Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Saturday, December 04, 2004 7:21 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] ZAP and IAX Trunks Hi Walid, Welcome to the list. Zap are the connections from ordinary pstn (or telco lines) to your digium hardware. IAX is an Asterisk protocol for incoming lines via IP from another asterisk PABX. Hope this helps. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid AzabSent: Saturday, December 04, 2004 5:42 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] ZAP and IAX Trunks HelloEveryone, I have recently come across these two terms. I am new at Asterisk and do appreciate your assistance in this. Some tools such as "astGUIclient" and "Asterisk Management Portal" require that the phone system be running Zap or IAX trunks as well as SIP devices. SIP devices are understadable but what about the other two? I am planning to use Cisco 7960/7940 IP phones. Thanks Walid ___ 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] Door buzzer.
Cian O'Sullivan [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) They have a pizza box server as their asterisk server with a T1 card. No more slots, so if I want to use the existing infrastructure I will need to build a second server with an FXO port. Kinda stupid having a second server just to open the door. If you need an FXO port and don't want to install a whole new server then you could consider an external device, such as a Sipura SPA-3000. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly 1.9.5 and 20041117 CVS HEAD -- IAX2 one way audio
Seb Auriol wrote: For the record, I experienced the same problem last week with only getting audio in one direction using Firefly 1.9.3.3934 and Asterisk 1.0, but only when Asterisk is bridging an IAX call with a TDM call. I didn't test it with IAX to IAX bridged calls, but the audio was fine (two way, although perhaps not both ways simultaneously) when Asterisk answered the call if I dialled a voicemail extension and left a message - I could hear the voicemail prompts and leave a message. Call quality was really poor though (very noisy). Please try a newer version -- 1.9.3 is getting quite old. Tim ___ 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] Recommendations for full featured phones
We are considering a replacement of a legacy PBX system (merlin). I am trying to figure out which phones would be best supported with the fullest set of features. Any recommendations? Sean ___ 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] Receptionist Phone
I have a customer interested in an * system, however she wants to ensure that the receptionist phone will display who is on the phone and who is not. It is an office of 10 people, and there are 12 PRI channels available. She is an older lady and does not want to use a web interface. Any suggestions? In other words, she wants to look at a device that indicates hook status of various extensions. I am guessing also that web interface extends to computer interface of any kind. Assuming the above, then why are they interested in Asterisk? If they like the ability to trunk between offices, for example, using inexpensive public Internet connections, Asterisk might have a place in this scenario, but from what you have said here, Asterisk is not the solution for their needs. Square pegs, round holes. They need a basic key system with a receptionist console. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Pocket PC over cell phone connection?
Anyone tried using a pocket pc with sjphone or x-ten over a cell phone connection? Uhh, good luck. Latency, lack of bandwidth... Nice idea, but I would stick with the cell phone when you're on the road. Or wait for WiMax service offering rollouts sometime in 2005. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Group sip definitions?
Been around * for over a year and I'm looking for a way to provide a simple set of group sip.conf phone definitions that can be used by multiple internal sip phones (as in 50/group for example). Something like: [All-Sales] type=friend username=* secret=sales99 callgroup=9 pickupgroup=9 [All-CustServ] type=friend username=* secret=cs88 callgroup=8 pickupgroup=8 If we _assume_ all sip phones are of the same type (eg, C7940's), is there a way to provide a single definition (such as All-Sales in the above example) that can be used by multiple phones keying off the 'secret=' to establish business oriented groupings? (Sort of like approaching plug-n-play at the sip phone level, using definitions specifically entered into each phone.) I fully understand how that might impact security, accountability, etc. It would appear that we now have sufficient * functions and macro capability to address extensions.conf definitions, checks for whether a voicemail box exists, etc, but I'm apparently brain-dead in finding a way to minimize the need to micro-manage individual sip def's. Any thoughts? Am I missing something or dreaming? Rich ___ 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] Receptionist Phone
I have a customer interested in an * system, however she wants to ensure that the receptionist phone will display who is on the phone and who is not. It is an office of 10 people, and there are 12 PRI channels available. You could look at the Snom 220 with its expanded call board. http://www.snom.com/snom220_en.php The LEDs on the SNOM phone's keys work very well with *. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] test
test ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco IAD2421 with Asterisk
All, I am posting this here to announce I have finally managed to get my Cisco IAD2421 to speak MGCP with Asterisk. Due to an acute lack of reading on the subject as searched on Google, I'm putting this out with the hope that it helps whomever should need to do this in the future. This should also apply to the IAD2420 and the other models in the line, but as I do not have access to those, they are untested. I have posted all my configurations and notes made during this effort in the wiki at http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+iad2420 (I know its not really a 'phone' but it seemed like the most appropriate way to list it.) Working and tested is the ability to call into the automated attendant and receive/process digits, as well as call another IAD channel and hold a conversation. A patch has been submitted at http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002982 (Asterisk bug ID 0002982) which improves the reliability of the IAD by hanging up calls when the IAD and Asterisk get out of sync. More details are in the bug report. My IAD2421 provides 16 analog pots channels via an Amphenol connector. It speaks MGCP with Asterisk and does minimal (if any) actual call processing internally. It is, however, also a full router with IPsec and SSH (if you have a recent enough version of the IOS and beefy enough hardware). Mine includes 1 Fast Ethernet port, 1 onboard Serial 0, and 1 T-1/PRI WIC. AFAIK, the WIC cannot be used for voice channels (data only). These devices are End Of Lifed as far as I know. Hope this helps! /BAK/ ___ 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] full duplex sound card
Hello, I have installed asterisk on fedora core 2. Can anybody suggest me a good full duplex sound card supported on linux. Thanks Varun ___ 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] X100P does not detect ringing
Michael Vogel schrieb: The X100P is working - partly. I can make outgoing calls. But the card has got a problem detecting incoming calls. Even in verbose mode I don't see any hint that the card detects a call. I debugging the module. At the moment it looks like there is a problem in the function wcfxo_receiveprep. The card does return something but the module doesn't really detect it as a ring. I cannot really imagine that I'm the first one who has this problem? Bye! Michael ___ 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] just testing please ignore
just testing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Gigabit Ethernet necessary?
[EMAIL PROTECTED] wrote: For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100 Mbps should be fine... The office will have 20 phones, with remote VoIP phones added to the mix later on. TIA, -Ron Ron, For what it costs, it is usually worth it to put a gig card in your server (a good one). Gigabit cards have newer and much better buffering and pci bus support. They are also much better at offloading processing from the system's CPU. You need to make sure that you have a good one. Because a crappy Gigabit card is probably not much better than a crappy 100mb card... I like Intel nics (both 100mb and 1000mb). Something supported by e1000 shouldn't be too expensive and usually will work pretty well with most all OS's. my $0.02 -- Kristian Kielhofner ___ 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] Using Pocket PC over cell phone connection?
snip Anyone tried using a pocket pc with sjphone or x-ten over a cell phone connection? Uhh, good luck. Latency, lack of bandwidth... Nice idea, but I would stick with the cell phone when you're on the road. /snip Latency is still a huge issue with the cell phone networks. DSL-Reports actually has a test you can run to check speed and show latency: http://text.dslreports.com/mspeed HTH, -Ron ___ 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] Recomended ISDN on Asterisk@home ?
Hi I have installed the http://asteriskathome.sourceforge.net/ with a Digium card with no problems, very good ! Now I want to install my Billion PCI ISDN card (HFC based) in TE mode. I get a little confused with Isdn4Linux, ZapHFC HIAX and the need to install Capi ! Please suggest best and easiest approach ? Thank you ! HB Norway ___ 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] Gossiptel with Asterisk?
I've done it in the past - maybe Gossiptel is just timing out? Most of these kinds of providers do, some more often than others On Sat, 4 Dec 2004 10:00:48 +, Ian Chilton [EMAIL PROTECTED] wrote: Hi, Has anyone got Gossiptel working with Asterisk? - I am having real problems getting it to register - i'm just getting timeout errors. Thanks --ian ___ 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] Two zaptel T1 cards: no clock from one
Oh, you got to be kidding me. :) I've removed zaptel, wct1xxp, wct4xxp completely, and reloaded in the same order as they are loaded, alternate different ways, etc. No help. After trying only wct1xxp with 12 channels and seeing alarms clear, something stray was left saying span 5 was 'UNCONFIGURED' (with only 1 span in zaptel.conf). So, I rebooted ro see if I could reset some phantom state, and viola! The damn thing is just fine now. Thanks for your help, all. I'll see if I can recreate this again and file a bug report, if so. ___ 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 dabbling...
Thanks for the reply Brian. I have a telecom background and just recently am getting heavily involved with VoiP at work... Guess I have to get the wife a new PC so I can use her's for playing PBX. I think I will go with Redhat 9 Linux. I will definately check out the providers you mention... Ray Brian Roy [EMAIL PROTECTED] wrote: On Sat, 4 Dec 2004 14:12:04 -0800 (PST), Ray Jender<[EMAIL PROTECTED]>wrote: Newbee hereRay,You should be fine with your setup. BSD can be a little finicky to getworking sometimes, but if you're familiar enough with it you will beOK. I have a P133 w/ 128mb ram running my home * box and I don't haveany problems with it. My wife doesn't even complain.For dialtone checkout any of the following. Nufone, Voicepulseconnect, broadvoice, voipjet. All of them have varying strengths. Youwill be able to connect to any of them over your broadband.Cuddle up to the wiki for a while. There is more information therethan you could possibly need. Asterisk is an adventure. Hope you'renot busy for the next couple months!-Chuji___Asterisk-Users ma iling list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard.___ 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] Two zaptel T1 cards: no clock from one
It is not important to have clocking between cards. Clocking is to make sure everyone knows how long a T1 frame should last and everyone is doing it exactly on schedule. The problem is that for the most part a frame is buffered. So while the DS0's are being processed, you don't have a new frame overwritting the old info before it's read or the DS0 buffer is read twice. If the timing between the two T1's is off that is a problem and that's called a slip. Either in reading the buffer, you get the same info twice for a certain voice channel or you miss reading and the buffer is overwritten before your card reads it. When you have two cards, you need to make sure that you are taking timing for the card from a good source. If you are connecting to a telco, they should all be good sources of timing and you shouldn't have any problems. Probably another concept that is confusing is that timing is not syncing the start and stop of the T1 frame, but really syncing the amount of time between the start and finish of that frame. Because the time to get the signal from point a to point b varies from the amount of time to get a signal from point a to point c, we cann't( and don't ) depend on the T1 frame starting at the same time. It's really about filling the buffer and having the correct amount of time to read it without overwriting it before it's read or reading it twice. Lyle - Original Message - From: Kevin Blackham [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, December 04, 2004 5:00 PM Subject: Re: [Asterisk-Users] Two zaptel T1 cards: no clock from one Yeah, proper crossover cable. I've eliminated all cabling issues with the T1 analyzer. I get a full and accurate pattern back when I test from the cable end where it would have been connected into the T100P, with the channel bank in loopback. The main symptom is that when I hook the analyzer directly to either the channel bank or the T100P, neither is providing clock. I could have the channel bank supply one, but I will have fax/modem calls bridged between the two PCI cards, so a common clock is best. The most disturbing thing is that the T100P, as the only card in a system, provided clock just fine. There was a thread last month in -dev about being unable to use common clock source across cards. Is this related? How can one cause zaptel to provide ref clock? Should I be seeing 1000 interrupts/sec on any and all TDM cards? On Fri, 03 Dec 2004 23:59:22 -0500, [email protected] wrote: The cable should be cross-connect 1-4, 2-5 each way. Is it? ___ 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] Is Gigabit Ethernet necessary?
snip [EMAIL PROTECTED] wrote: For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100 Mbps should be fine... The office will have 20 phones, with remote VoIP phones added to the mix later on. If you are using a switch, collisions a pretty much a non-issue, unless you have enough traffic to saturate a port to the server. Latency is also not helped any significant amount, since you still have a 100Mbit link in the path between the phone and Asterisk. In other words, for that application, it likely will not make any difference at all. If it's cheap to do, and the server will also be doing any file serving duties, then it would be a nice insurance policy against a single user swamping the server's port. /snip Sounds good to me. The server will be dedicated to Asterisk, so no worries about other applications (unless I move the config to a database which down the line could be very likely). Regards, -Ron ___ 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] Problem with Grandstream bt100
I was registering a bt100 with asterisk but can't do it and then i restart the phone , then phone never come back the lcd is lit but in blank, and the four internal leds are flashing. Did you set a tftp server on your phone? If you did then most likely the phone was downloading the firmware when the leds were flashing. If you pulled the plug then you might have destroyed your firmware and the phone along with it. Holden ___ 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 + chan_sip2 + sipproxd + sipgate
Hi, i have an asterisk server behind a masquerading firewall and trying to register to sipgate.de. I use for outbound connections chan_sip2 and on the firewall sipproxd as outound proxy, but it doesnt work. Could anybody help me? My firewall has ip addresses: 172.18.48.151 [my dyndns] My asterisk server has ip address: 172.18.48.254 My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 ;externip = [my dyndns] context = sip_in srvlookup=yes canreinvite=no disallow=all allow=gsm allow=ilbc allow=alaw allow=ulaw allow=all outboundproxy=172.18.48.251 outboundproxyport=5060 localnet=172.18.0.0/255.255.0.0 register = [sipgate authid]:[sipgate password]:[sipgate [EMAIL PROTECTED]/s ; [sipgate] type=friend username=[sipgate authid] fromuser=[sipgate authid] fromdomain=sipgate.de host=proxy.de.sipgate.net secret=[sipgate password] context=sip_in insecure=very nat=yes My sipproxd.conf: # Minimal config only. Derived from siproxd.conf.example # and changed: if_(in|out)bound, user, (regist|pid)file if_inbound = vlan2 if_outbound = ppp0 daemonize = 1 user = root registration_file = /var/run/siproxd_registrations pid_file = /var/run/siproxd.pid rtp_proxy_enable = 1 rtp_port_low = 12000 rtp_port_high = 13000 rtp_timeout = 300 mask_host = 172.18.48.251 masked_host = [my dyndns] ___ 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 dabbling...
Ray Jender wrote: Newbee here I would like to play around with Asterisk a little. First, I need to prepare a server with FreeBSD. It's a PII 433mHz/256mb box. Good enough? Then install Asterisk. I have a broadband (cable) internet presence. Could I do anything with this connection and Asterisk? Thanks, Rayasterisk Ray, I hate to say this (I am a huge FreeBSD fan), but I believe that each OS has it's own strengths. While FreeBSD isn't any better or worse than Linux for Asterisk, Linux was the platform that it was originally developed on. It sounds like you are new so I will suggest that you stick with Linux for now and enjoy more support options, better hardware support, and more documentation. I have run Asterisk on both (even inside a FreeBSD jail) and I will say that I prefer to run it in Linux because as of now it just works better. My web servers, mail servers, etc, etc, etc. can run FreeBSD because I happen to like FreeBSD for those tasks. But not for running * (as of now, that could change...) That hardware should be fine, but then again I don't even know what you will be doing with it. There are people that run * on P133's. But like anything else, don't expect it to be able to work magic just because it is Linux and OSS. Hardware limits are still hardware limits. I would say though, that for most of the common stuff that you will want to play around (dabble) with, this machine sounds fine (some would say more than fine). I have run it on much less http://www.krisk.org/astlinux/ As for cable internet, it all depends. How much bandwidth do you have, are you behind NAT? What kind of packet loss/latency/jitter do you typically experience? If I were you I would just give it a shot and see how it works! P.S. - use kernel 2.6 if you can -- Kristian Kielhofner ___ 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 dabbling...
Hi Brian, Just a quick question. Do I need any other hardware if I want to use Asterisk over my cable broadband connection? Would I be using my existing NIC card in the PC? Are there any "how to" documents available for what I am doing? Thanks for your help. Ray Brian Roy [EMAIL PROTECTED] wrote: On Sat, 4 Dec 2004 14:12:04 -0800 (PST), Ray Jender<[EMAIL PROTECTED]>wrote: Newbee hereRay,You should be fine with your setup. BSD can be a little finicky to getworking sometimes, but if you're familiar enough with it you will beOK. I have a P133 w/ 128mb ram running my home * box and I don't haveany problems with it. My wife doesn't even complain.For dialtone checkout any of the following. Nufone, Voicepulseconnect, broadvoice, voipjet. All of them have varying strengths. Youwill be able to connect to any of them over your broadband.Cuddle up to the wiki for a while. There is more information therethan you could possibly need. Asterisk is an adventure. Hope you'renot busy for the next couple months!-Chuji___Asterisk-Users ma iling list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more.___ 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] Hardware PSTN Gateways?
I am thinking about setting up an asterisk PBX system for my company. But since I can't be at all the locations all the time I am setting up an automatic backup system where if the backup detects that the primay is down it takes over the IP so calls can be made once more. For this reason I want to setup a seperate HARDWARE PSTN Gateway. Are there any equiptment that can be plugged into the network, connected to a PSTN line and just act as PSTN gateway. It needs to handle both incoming and outgoing calls. And preferably handle more than one pstn line per box. TIA, Tomoki ___ 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] Door buzzer.
I have in my house a device called DoorBell Fon, which connects to an FXO port. When a visitor presses the button on the intercom, Asterisk will see an incoming call. You can configure your dialplan to react as you wish. You can also purchase a lock controller which will solve the problem of opening the door. You can buy it from www.smarthome.com, or directly from the manufacturer. Aren't there USB FXO adapters that you could use? Fernando On Dec 4, 2004, at 6:00 PM, Cian O'Sullivan wrote: Hello, I have a customer who has their front door integrated to their current phone system. If someone presses the buzzer, the secretaries phone will ring, and she can talk to the person at the door. By pressing ** she can release the door. Anyone have any sort of integration like this. Are there IP devices anyone is using? They have a pizza box server as their asterisk server with a T1 card. No more slots, so if I want to use the existing infrastructure I will need to build a second server with an FXO port. Kinda stupid having a second server just to open the door. Any suggestions? Cian ___ 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] chan_misdn and Dynalink IS64PH ISDN
--- bagattin jerome [EMAIL PROTECTED] a écrit : --- Simon Richter [EMAIL PROTECTED] a écrit : Hi, TE Stack No Upper ID init_stack: File exists You need to set the layermask when loading the card driver. For a TE port, use 15 (layer 0-3) and for an NT port, use 3 (layer 0-1). Simon Thanks, I add layermask in my modprobe script : /sbin/modprobe --ignore-install w6692pci protocol=2 layermask=3 Now I have another error : Init. Stack on port 1 TE Stack No lower Id init_stack: File exists In syslog : kernel: MISDN free_device: entitylist not empty What can I do to resolv that ? thanks Jerome I have make a error, if I had layermask in insmod for w6692pci capiinfo don't see anything !! How can I set the layermask parameters ? Thanls Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ Avec Yahoo! faites un don et soutenez le Téléthon en cliquant sur http://www.telethon.fr/030-Don/10-10_Don.asp ___ 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 and Cisco IP Phones
What do you suggest then Brian? Thanks Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, December 04, 2004 9:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones Let me CLARIFY for those that might not get what I ment.. DO NOT RECOMMEND SCCP unless you have actually installed and used it. Its crap... SIP is what you want if you use a cisco phone with asterisk. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, December 04, 2004 1:33 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones Pfft ya right if you want half assed supported channel drivers. Use SIP. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Keith O'Brien Sent: Saturday, December 04, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones No you don't have to use SIP. You can also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: Walid Azab [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ 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] Sip Channels Left Open
Hi, If I do a sip show channels - I seem to be getting channels left open after calls have ended - any ideas why? I thought at first it was my Sipura SPA-3000 and that Asterisk was not detecting that i've hung up. However, after more testing, it seems to be just on Gossiptel calls - I tried a few of my other sip providers and the channels stay open after the call has ended but then dissapear after about 30 seconds. With Gossiptel calls - the channels just seem to stay open forever (or at least for a long time causing me to get errors about running out of rtp ports). I tried a software sip client and this has the same behaviour - channels left open for about 30 sec after sip calls but left open forever on gossiptel calls. Any ideas? Thanks! --ian ___ 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] Ring all Configured Extension
On Thu, 2 Dec 2004, Eric Rees wrote: Where only talking about 100 extensions. That is a lot to hard code by hand. Just use app_queue and define a list of members as the SIP extensions. It is a lot easier to maintain the queues.conf file than to worry about adding 100 extensions into your dial-plan. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Thursday, December 02, 2004 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Ring all Configured Extension Why are you afraid of that suggestion? Matthew - Original Message - From: Eric Rees [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 02, 2004 10:56 AM Subject: RE: [Asterisk-Users] Ring all Configured Extension I was afraid that someone would suggest that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Thursday, December 02, 2004 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Ring all Configured Extension exten = 4000,1,Dial(SIP/3001SIP/3002SIP/3003...on and on, 30, t) Matthew - Original Message - From: Eric Rees [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 02, 2004 8:56 AM Subject: [Asterisk-Users] Ring all Configured Extension I don't know if the is possible on not. I would like to know the easiest way to ring all extensions in the sip.conf file for intercoms. I have phone to phone intercom working. ___ 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 PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] List's quiet or down?
Is it just me or are there problems? The list has just shutdown over the last 24 hours :( David ___ 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 dabbling...
Hello: On my ppoint of view, it is a good hardware to start But the performance depends of ... - How many stations you will have calling a the same time, to the "outside" world ? - What kind codecs you will use ? Take a look to: http://www.voip-info.org/wiki-Asterisk+Hardware Or simple go to http://www.voip-info.org/tiki-index.php?page=Asterisk or www.asterisk.org And you will find ALL info that you need to start Hope this help ! ---Ing. Julio Alvarez TejeraUnix Trends*BSD, Solaris Linux---"extremely stable systems" - Original Message - From: Ray Jender To: [EMAIL PROTECTED] Sent: Saturday, December 04, 2004 4:12 PM Subject: [Asterisk-Users] asterisk dabbling... Newbee here I would like to play around with Asterisk a little. First, I need to prepare a server with FreeBSD. It's a PII 433mHz/256mb box. Good enough? Then install Asterisk. I have a broadband (cable) internet presence. Could I do anything with this connection and Asterisk? Thanks, Rayasterisk Do you Yahoo!?Read only the mail you want - Yahoo! Mail SpamGuard. ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Receptionist Phone
Cian O'Sullivan wrote: Hello, I have a customer interested in an * system, however she wants to ensure that the receptionist phone will display who is on the phone and who is not. It is an office of 10 people, and there are 12 PRI channels available. She is an older lady and does not want to use a web interface. Any suggestions? Cheers Cian ___ 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 Snom 220 ___ 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] Door buzzer.
On Sat, 4 Dec 2004, Cian O'Sullivan wrote: They have a pizza box server as their asterisk server with a T1 card. No more slots, so if I want to use the existing infrastructure I will need to build a second server with an FXO port. Kinda stupid having a second server just to open the door. If the device is only a buzzer, can't you do anything fancy on the comport, with hardware and an event poll? Or if it is a phone device maybe an Iaxy can do the trick? Stefan de Konink ___ 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] Receptionist Phone
Use a snom phone (220) with a side car, These work great with the hint priority, What you are describing is a DSS/BLF button. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cian O'Sullivan Sent: Saturday, December 04, 2004 6:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Receptionist Phone Hello, I have a customer interested in an * system, however she wants to ensure that the receptionist phone will display who is on the phone and who is not. It is an office of 10 people, and there are 12 PRI channels available. She is an older lady and does not want to use a web interface. Any suggestions? Cheers Cian ___ 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] TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment
Richard Scobie wrote: Rich Adamson wrote: The tdm card does have some unusual issues that appear to be driver oriented, but there are lots of folks using the card in production. Usually in situations where the client knows how to and tolerates having to reload drivers and/or reboot his PBX periodically... Even if the client knows that, even if he/she would build a cron with asterisk -rx stop; rmmod wctdm etc.. Imagine if that would be happening in the middle of the day.. breaking all phone conversations... ___ 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] Door buzzer.
- Original Message - From: Cian O'Sullivan To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, December 04, 2004 7:00 PM Subject: [Asterisk-Users] Door buzzer. Hello, I have a customer who has their front door integrated to their current phone system. If someone presses the buzzer, the secretaries phone will ring, and she can talk to the person at the door. By pressing ** she can release the door. Anyone have any sort of integration like this. Are there IP devices anyone is using? They have a pizza box server as their asterisk server with a T1 card. No more slots, so if I want to use the existing infrastructure I will need to build a second server with an FXO port. Kinda stupid having a second server just to open the door. Any suggestions? Cian I assume you could use a Grandstream 286 or two devices would be even better. Maybe a GS 102 programmed with autodial to the secretary and depending on the secretary's phone you could program a speed dial to the GS 286 which in turn will send ring voltage to the relay that unlatches the door. Make sure that exentension is only available from the internal dialplan. Also for some really cool wow effect, make it available from your incoming context with an authenticate line. Now employees can open the door with their cell phones after entering a pin. Thanks, Steve Totaro ___ 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] Sip no voice
Hi Serge - The connection works fine in my internal network, only outside callers have no voice. Thanks for the firefly config.Can you provide me your sip.conf from the machine you are using to run asterisk? It might be that the sip.conf file is not allowing your asterisk machine to connect with the phones using the right codecs. What does it say on the asterisk console when you try to dial one phone from another? If you don't see anything, try running asterisk with: asterisk -vvgc Another thing to try would be other softphones. I've never used firefly before, but have had success with both SJPhone and Xlite. Thanks, Noah -Original Message-From: Noah Miller [mailto:[EMAIL PROTECTED] Sent: mercredi 1 décembre 2004 14:56To: Asterisk Users Mailing List - Non-Commercial DiscussionCc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Sip no voice Hi, What can it be when I can establish a connection between two Softphones but no voice is transfered ? thnx Hugo, It could be a codec problem, or many other things - can you provide more detail? What softphone is it? What codec(s) are you trying to use? If it's a SIP softphone, what's your sip.conf, extensions.conf, etc? Thanks, Noah ___ 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 + chan_sip2 + sipproxd + sipgate
Hi, i have an asterisk server behind a masquerading firewall and trying to register to sipgate.de. I use for outbound connections chan_sip2 and on the firewall sipproxd as outound proxy, but it doesnt work. Could anybody help me? My firewall has ip addresses: 172.18.48.151 [my dyndns] My asterisk server has ip address: 172.18.48.254 My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 ;externip = [my dyndns] context = sip_in srvlookup=yes canreinvite=no disallow=all allow=gsm allow=ilbc allow=alaw allow=ulaw allow=all outboundproxy=172.18.48.251 outboundproxyport=5060 localnet=172.18.0.0/255.255.0.0 register = [sipgate authid]:[sipgate password]:[sipgate [EMAIL PROTECTED]/s ; [sipgate] type=friend username=[sipgate authid] fromuser=[sipgate authid] fromdomain=sipgate.de host=proxy.de.sipgate.net secret=[sipgate password] context=sip_in insecure=very nat=yes My sipproxd.conf: # Minimal config only. Derived from siproxd.conf.example # and changed: if_(in|out)bound, user, (regist|pid)file if_inbound = vlan2 if_outbound = ppp0 daemonize = 1 user = root registration_file = /var/run/siproxd_registrations pid_file = /var/run/siproxd.pid rtp_proxy_enable = 1 rtp_port_low = 12000 rtp_port_high = 13000 rtp_timeout = 300 mask_host = 172.18.48.251 masked_host = [my dyndns] ___ 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] help
help ___ 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] Bluetooth with *
Martin List-Petersen wrote: Check http://www.crazygreek.co.uk/content/chan_bluetooth, but it's still in heavy development. Far from finished. Isn't there also a module to allow location tracking via bluetooth, that is, the room you are in is triangulated via bluetooth and your calls are routed to the nearest phone? I'm sure I remember reading something about it at one point... ___ 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] Email to Fax?
I've read about Fax to Email, but is there such a beast as email to fax? If not, what do people use to take care of outbound faxing? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP Address?
Hi!
[default]
exten = ian,1,Dial(SIP/spa3k_line1,10)
exten = ian,2,Voicemail(u4)
exten = ian,3,Hangup
Is there any way to get such calls coming into a dedicated context,
rather than default?
Use gotoif() and the variable ${SIPDOMAIN}
Cheers, Philipp
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Re: [Asterisk-Users] TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment
Thank you for your answer. Now I've figured that one of the FXO modules on the card may be defective. Whenever I plug in telco line in it - that line will be like shortened (if you pick up parallel telephone, the dial tone will be heard weaker than usually). So the FXO module is always in Offhook state, unable to dial out, unable to detect rings. Reboot and Power off/Power on did not help. Any suggestions? Might be just my luck.. just my luck. If you're sure that jack is a fxo module, then I'd call digium support. ___ 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] Receptionist Phone
Use a snom phone (220) with a side car, These work great with the hint priority, What you are describing is a DSS/BLF button. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cian O'Sullivan Sent: Saturday, December 04, 2004 6:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Receptionist Phone Hello, I have a customer interested in an * system, however she wants to ensure that the receptionist phone will display who is on the phone and who is not. It is an office of 10 people, and there are 12 PRI channels available. She is an older lady and does not want to use a web interface. Any suggestions? Cheers Cian ___ 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] System hardware requirements for *
What would be a the minimum hardware requirements for a small asterisk pbx that would only have 2 pots lines coming in(2 fxo ports)but with 4 extensions(4 fxs ports) And enough space to hold up to a month of vmail for those 4 extensions/users? Heck what arethe typical hardware requirements of * anyways? I can't seem to find this on the website. I know it all depends on the situation,but what's typical? Thanks, Brent ___ 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 and Cisco IP Phones
Guys, obviously there is an argument about SIP vs SCCP when it comes to using Cisco IP Phones with Asterisk. I am not really sure which way to go. Probably I will go with SIP now unless you guys do recommend not using it. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, December 04, 2004 9:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones Let me CLARIFY for those that might not get what I ment.. DO NOT RECOMMEND SCCP unless you have actually installed and used it. Its crap... SIP is what you want if you use a cisco phone with asterisk. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, December 04, 2004 1:33 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones Pfft ya right if you want half assed supported channel drivers. Use SIP. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Keith O'Brien Sent: Saturday, December 04, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones No you don't have to use SIP. You can also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: Walid Azab [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ 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] ANALOG FXO ZAPTEL WCFXO WCTDM module issues seen with intermittent analog lines
Hello, I have found a bug, I think in the way TDM400P cards handle FXO interface disconnect/re-connect problems. Normally I do keep all the wires connected from my CO / PABX quite securely, but I had a need to re-route the cable from one side of the desk to another, and I simply disconnected the RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY SCRATCHY AUDIO CONNECTIONS WHEN USING THE FXO PORT. Incoming calls were erratic, outbound calls were almost unuseable, dialled digits were almost unrecognizable. Basically after some difficult troubleshooting the fix was: before disconnecting cable kill asterisk process remove WCTDM module remove WCFXO module remove ZAPTEL module and then, reconnect cable, and then, install ZAPTEL, WCTDM, WCFXO, start asterisk once again. Apparently during operation of the zaptel driver, disconnect of the cables to the ports is not recommended. I can replicate this condition easily, and if other users have any trouble with their analog ports due to the fact that their connections are flaky after some change (while asterisk is running) I would love to know how they coped with it. -samudra --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.804 / Virus Database: 546 - Release Date: 11/30/2004 ___ 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] Two zaptel T1 cards: no clock from one
Since Span 5 is a channelized T1 make sure the last two parameters match the settings for the chan bank(esf b8zs). You cann't remove timing from a T1. You can only make YOUR card take timing from the incoming digital signal, take timing from another T1 on the same card or supply it's own internal timing. (these cards do not pass timing between cards from what I have read here.) My gut reaction is that something else is wrong. I also noticed in your debug output that span 1 through 4 were represented, but span 5 does not show up. What about ztcfg and zttool outputs? Lyle - Original Message - From: Kevin Blackham [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, December 03, 2004 6:57 PM Subject: [Asterisk-Users] Two zaptel T1 cards: no clock from one List, I have a TE410P (T1 mode, all PRI) and a T100P (fxoks, for fxs channel bank). I cannot seem to get the T100P to send any clock to the channel bank. I prefer that it use the same clock source as the TE410P, but it doesn't matter if it's not in sync just as long as it's there. The TE410P is configured 3x pri_cpe, 1x pri_net. The three cpe go to XO Sonus switch, the net to legacy PBX. Clock is received from telco, old PBX receives clock from zaptel card, everything's green there, but the other card, the T100P, seems to not send any timing at all, as verified by our T1 analyzer, and is persistently in red alarm. In fact, even if I stick a loopback plug in the T100P, the alarm persists (loopback causes a result in the TE410P). The T100P and channel bank were just pulled from another working * box, and the configuration is nearly identical, except it was the only T1 interface. System: Supermicro dual Xeon 2.4, both cards on same PCI bus. Cards: one T100P, one TE410P. Config: spans 1-4 for quad card (module loaded first), span 5 is single port card Channel bank: Access Bank II, 12 FXS Info dumps (some snipped for brevity) lspci (snipped, these are the only devices on bus 5): :05:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface :05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01) cat /proc/interrupts (odd, shouldn't the T100P be generating 1000 ints/sec?): CPU0 CPU1 CPU2 CPU3 0: 20727 0 18569488 0IO-APIC-edge timer 9: 0 0 0 0 IO-APIC-level acpi 28: 177705 0 0 0 IO-APIC-level eth0 29: 9282 0 0 0 IO-APIC-level eth1 72: 35845 0 0 0 IO-APIC-level dpti0 100:102 0 0 0 IO-APIC-level t1xxp 104: 18342391 0 0 0 IO-APIC-level t4xxp lsmod: Module Size Used by wct1xxp17568 0 wct4xxp70048 0 zaptel226436 222 wct1xxp,wct4xxp e1000 87348 0 crc_ccitt 3072 1 zaptel zaptel.conf: span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,3,0,esf,b8zs span=4,4,0,esf,b8zs span=5,0,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 bchan=49-71 dchan=72 bchan=73-95 dchan=96 fxoks=97-108 #fxoks=109-120 loadzone = us defaultzone=us asterisk/zapata.conf: [channels] language=en echocancel=yes echocancelwhenbridged=no echotraining=yes echotraining=800 immediate=no ;--pstn-- context=from-pstn signalling=pri_cpe switchtype=dms100 group = 1 channel = 1-23,25-47,49-71 ;--pri to pbx-- signalling=pri_net switchtype=dms100 group = 3 channel = 73-95 ;--channel bank-- context=fax+modem signalling=fxo_ks channel = 97-108 a snippet from dmesg: ACPI: PCI interrupt :05:03.0[A] - GSI 104 (level, low) - IRQ 104 Found TE410P at base address f8401000, remapped to f9b98000 TE410P version c01a009b FALC version: 0005, Board ID: 00 registers snipped TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx ACPI: PCI interrupt :05:02.0[A] - GSI 100 (level, low) - IRQ 100 Framer: DS21552, Revision: 3 (T1) Found a Wildcard: Digium Wildcard T100P T1/PRI Registered tone zone 0 (United States / North America) TE410P: Span 1 configured for ESF/B8ZS SPAN 1: Primary Sync Source TE410P: Span 2 configured for ESF/B8ZS SPAN 2: Secondary Sync Source TE410P: Span 3 configured for ESF/B8ZS SPAN 3: Tertiary Sync Source TE410P: Span 4 configured for ESF/B8ZS SPAN 4: Quaternary Sync Source Using ESF/B8ZS coding/framing Calling startup (flags is 4099) ___ 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] Broadvoice outbound 404 error
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sat, Dec 04, 2004 at 06:03:46PM -0600, Brian Roy wrote: On Sat, 4 Dec 2004 17:22:38 -0500, Reid Forrest [EMAIL PROTECTED] wrote: Is anyone else experiencing 404 errors on outbound dial with Broadvoice? I've followed the instructions posted by Broadvoice to configure sip.conf, and inbound calling works fine. Every time I try to dial out, I get a 404 Not Found error. [bv-home] type=peer host=proxy.dca.broadvoice.com Change the above line to host=sip.broadvoice.com Give that a try. -Chuji Chuji I think I am going to have to kill you. sip.broadvoice.com isn't proxy.dca.broadvoice.com meaning it will probably be sending the call half way across the country. Brian -- Go into your hosts file and get the ip of proxy.dca.broadvoice.com and set its like this ip sip.broadvoice.com then do as chuji suggested - -- Dan -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBs4UtF6i3K/AxoQERAvZ5AKC3z01CCklWxCFIt/Xgog2bdGKLTACbBrjV JHnwJGhDYM7KrVBEb3OLMD0= =fEls -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] full duplex sound card
Hello, I have an onboard sound AC97. Howto to find if my sound is full duplex ? And if my sound is not full duplex then please recommned me a good full duplex sound card that is supported on linux. Thanks varun ___ 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] Budgetone 100 Caller ID
At 06:24 PM 12/4/04, you wrote:
Greg - Cirelle Enterprises wrote:
Hi,
Is there an * configuration that will allow the BT100 to
display the numeric callerid instead of the broken
text?
exten = extension,priority,SetCIDNum(${EXTEN})
Doug
Thanks Doug, will try that
Greg
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Re: [Asterisk-Users] X100P does not detect ringing
Michael Vogel schrieb: The X100P is working - partly. I can make outgoing calls. But the card has got a problem detecting incoming calls. Even in verbose mode I don't see any hint that the card detects a call. Now it works. I changed the following items in the file wcfxo.c: #define PEGTIME 1000 * 8 #define PEGCOUNT 0 static int opermode = 1; With these values it seems to work. (For the archieve: These values are working for me in germany) Bye! Michael P.S.: The list seem to have problems at this time. I hope the mails I posted aren't lost. I haven't received mails since 1:06 GMT+1. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SJPhone SIP Tab
Hi Norman, I played with this for ages also. I think there is a small step missing from the wiki that needs explainantion. Prior steps in the SJPhone setup: 1/ click on the Options button 2/ go to profiles tab. 3/ click on 'New' 4/ create a new profile called 'asterisk' with profile type 'Calls through SIP proxy' 5/ use this profile for your asterisk connection follow the wiki from there. :) hope this helps, I edited the Wiki to show these steps in case somebody else out there has the same problem. I hope this is OK with everybody? Cheers, Mick Norman Zhang [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone. However, I cannot find the SIP tab. Would someone please give me a few pointers? The screen capture can be seen at URL below http://www.dslreports.com/forum/remark,12022987~mode=flat Regards, Norman Zhang ___ 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] PRI configuration problem
We've been working for the past 2 weeks to get a new V400P working with our PRIs from the telephone company. We're trying to get the Asterisk server setup as a VoIP gateway for SIP and AIX. We can make SIP-SIP calls, but all calls from or to the PRI fail. This is the applicable entries from the Asterisk log (configuration files follow) for a call coming from the PSTN on the PRI. I believe that the cause of the error is related to the line, Ring requested on unconfigured channel 0/23 span 1. But as far as I can tell, the channels are all configured. Protocol Discriminator: Q.931 (8) len=45 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: SETUP (5) [04 03 90 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 97] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 23 ] [1e 02 8a 01] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network beyond the interworking point (10) Ext: 0 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] [6c 0b 80 36 31 38 34 33 34 31 30 30 30] Calling Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation permitted, user number not screened (0) '6184341000' ] [70 0b a1 36 31 38 34 33 34 31 35 30 30] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '6184341500' ] -- Making new call for cr 1 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) Dec 6 04:19:43 WARNING[4891]: Ring requested on unconfigured channel 0/23 span 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 ac] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Requested channel not available (44), class = Network Congestion (2) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Zaptel.conf --- span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 bchan=49-96 loadzone = us defaultzone=us = Zapata.conf --- [trunkgroups] trunkgroup = 1,24,48 spanmap = 1,1,1 spanmap = 2,1,2 spanmap = 3,1,3 spanmap = 4,1,4 [channels] group=1 callgroup=1 pickupgroup=1 context=from-pstn switchtype=national signalling=pri_cpe channel = 1-23,25-47,49-96 language=en usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 immediate=no callerid=asreceived echocancel=yes echocancelwhenbridged=yes echotraining=400 Extensions.conf --- [general] static=yes writeprotect=yes [from-pstn] exten = 6184341500,1,Dial(SIP/6184341500,20) exten = 6184341500,2,Voicemail2(u6184341500) exten = 6184341500,102,Voicemail2(b6184341500) exten = 6184341500,103,Hangup exten = 4341500,1,Dial(SIP/6184341500,20) exten = 4341500,2,Voicemail2(u6184341500) exten = 4341500,102,Voicemail2(b6184341500) exten = 4341500,103,Hangup [from-internal] exten = _NXX,1,Dial(Zap/g1/$(EXTEN)) exten = _NXX,2,Congestion === Sip.conf [6184341500] callerid=GlobalEyes 6184341500 canreinvite=no context=from-internal dtmfmode=rfc2833 host=dynamic mailbox=xxx nat=yes port=5060 secret=xxx type=friend username=xxx allow=all ___ 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] Planet BRI TA will work ?
Dear all i am new to Asterisk i want to configure Planet TA ( Terminal Adapter ) for outgoing calls which module to use capi or linux4isdn to be used ? Thanks and Regards Talha ___ 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 and Cisco IP Phones
Thanks Keith..could you please send me any useful info on SCCP usage and how I can use it with Cisco IP Phones. Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Keith O'BrienSent: Saturday, December 04, 2004 8:57 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones No you dont have to use SIP. You can also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: "Walid Azab" [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset="us-ascii" Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ 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] Polycom IP500
Does anyone have a location to download the latest Polycom firmware etc? Other than the extranet site, because I am not a reseller, there fore I have no login. [minirant] And shouldn't end users be granted access to this kind of thing anyway? Geeze [/minirant] Thanks, Chris Cherry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.289 / Virus Database: 265.4.5 - Release Date: 12/3/2004 ___ 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] Why, why, why???
On Fri, 2004-12-03 at 16:54 -0500, Ferguson, Michael wrote: [incoming] exten = 321XXX,1,Goto(incoming,s,1) Afaik all regex numbers should start with an underscore so that should read _321XXX I guess. [snip] SIP.CONF [general] port=5060 bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) externip=XXX.XXX.XXX.XXX localnet=192.168.131.0 localmask=255.255.255.0 context=incoming tos=lowdelay disallow=all allow=ulaw context=invalid You have a context in here twice. That looks like one too many. Regards, Patrick ___ 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] Is Gigabit Ethernet necessary?
[EMAIL PROTECTED] wrote: For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100 Mbps should be fine... The office will have 20 phones, with remote VoIP phones added to the mix later on. The reason to chose a Gigabit Ethernet card has nothing to do with bandwidth - (most of?) these card use some sort of interrupt mitigation technique which takes a hell lot of load off of the processor for dealing with interrupts. VoIP traffic, with it's typical many small packets, is very susceptible to causing interrupt live lock on servers and routers and interrupt mitigation scheme (or even polling, but that's rare) makes a real change in performance. Having said that, there are 100Mb cards that do interrupt mitigation as well (for example AFAIK the Intel e100 cards) and there are drivers that implement interrupt mitigation at the software level (customized drivers for the tulip chip set based cards and the Linux NAPI framework). However, it is simply much easier to just grab a Giga card then research which 100Mb chip and which driver you need to get ;-) Hope this helps, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] G.729 algorithm?
hi all according to what I've found out this far, the G.729 patent seems not valid in a broad range of countries. so... does anyone know where I can find the algorithm? roy ___ 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] Dial D option not working?
For some reason I cannot get the 'D' option to send dtmf after connect. This doesn't work exten = _XXX, 1, Dial(Zap/r3,10,d(300) ) This does: exten = 300, 1, Dial(Zap/r3,10,M(to-300) ) [macro-to-300] exten = s,1,SendDTMF(300) Of course, what I really need to send is not 300, but $EXTEN but since I am running 1.0 and do not have the patch that allows macro arguments I cannot pass the exten into the macro... The only idea I can think of is to stuff it into a variable, but I would worry about race conditions. Mark ___ 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] NOTICE[507921]: app_dial.c:742 dial_exec:Unableto create channel of type 'Zap'
Is this the only device on IRQ 12? What does ztcfg -vvv show? Lyle - Original Message - From: U. Abdullah Sheikh [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Wednesday, December 01, 2004 9:46 AM Subject: Re: [Asterisk-Users] NOTICE[507921]: app_dial.c:742 dial_exec:Unableto create channel of type 'Zap' Hi Adamson, Thanks for such a comprehensive answers. Below is some more data for your feedback. I tried all, but it is still not working. Any comments and advise based on below data? 0. The System is in Singapore. 1. I have an X100P Generic Clone Card bought over from eBay. 2. lspci output: 00:0e.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Intel Corp.: Unknown device 0003 Flags: bus master, medium devsel, latency 32, IRQ 12 I/O ports at ec00 Memory at ef001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 3. lsmod output: Module Size Used by wcfxo 12448 0 zaptel241028 1 wcfxo crc_ccitt 1985 1 zaptel 4. /usr/sbin/zaptel/zttool output: I see the output below: Zaptel Tool (C)2002 Linux Support Services, Inc. â⤠Zapata Telephony Interfaces âââ â â â Alarms Span â â OK Generic Clone Board 1 â â â â â ââ⤠Generic Clone Board 1 ââ ââ âCurrent Alarms: No alarms. â âSync Source:Internally clocked â âIRQ Misses: 0 â âBipolar Viol: 0 â âTx/Rx Levels: 0/ 0 â âTotal/Conf/Act: 1/ 1/ 0 â Span 1: 1 total channels, 1 configured F1=Details F10=Quit 5. the show modules from asterisk CLI ... output below: chan_zap.so Zapata Telephony w/PRI 0 6. Zapata config is pasted below: [channels] relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=yes context=bell signalling=fxs_ks callerid=asreceived channel = 1 thanks regards Original Message Follows From: Rich Adamson [EMAIL PROTECTED] Would you tell us what country this system is in? The zap show channels should look something like: phoenix*CLI zap show channels Chan Extension Context Language MusicOnHold pseudo inbound-bus-x10 en default 1 inbound-bus en default and the 'zap show channel 1' should fill your cli screen with relevent data. So, yes you have a problem with the zap channel, but with the data included in your posting there isn't enough info to point to an exact cause. From the linux command line, do a 'lspci' and look for something that says Tiger Jet. If you don't see something related to the x100p, then your system isn't recognizing the x100p. (I'm assuming this _is_ a digium x100p and not one of the knockoffs.) From the linux command line, do a 'cat /proc/interrupts' and look for the x100p driver (wcfxo if memory serves correctly). Is it there? Change directory to /usr/src/zaptel and do a './zttool' from the command line. Do you see the x100p listed? From the linux command line, do a 'lsmod'. Is the wcfxo and zaptel drivers listed? Does the zaptel entry have a [wcfxo] to the right side of the line? From an asterisk cli, do a 'show modules'. Do you see something like: chan_zap.so Zapata Telephony w/PRI If you see acceptable entries for all of the above, then it would appear something is very wrong with your /etc/asterisk/zapata.conf file. Don't know what, but could be spaces inserted where there shouldn't be, control characters embedded that can't be seen, or whatever. Worst case, rename that file and create a new one ensuring all entries are entered correctly. Rich Hi Rich Adamson, Thanks for your valuable reply. The telco line is connected and working properly. The phone number is also correct (see the debug messages). 1. I suspected it may be SIP - SIP issue, which might be causing SIP to PSTN dialout problem. 2. Is there any command, which I
[Asterisk-Users] Re: Is Asterisk-users down?
David, I found your post on the Digium archives because I too have noticed that the flow of traffic on the list has stopped for the past 24 hours or so. I have replied to many existing threads and started new ones, only to not see my new messages. I take it from your recent post that you too have experienced this. Are you still not getting anything? Nothing. Not a damm thing :( So there must be a block somewere on the outbound side of things if My email made it to the archives. Hmm wonder what is going on? Thanks for confirming that it is not only me that is having a challange. David Thanks. -- Kristian Kielhofner ___ 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] Door buzzer.
I have the same problem setup with one of our customers, but I have a different problem. I have Grandstream ATA 486 connected to clients doorphone system and clients *. I have two problems: first, when someone is calling from the doorphone, ATA doesn't recognize the called number correctly. 2 or tries out of 10, ATA get's the number wrong. I have tried all kinds of DTMF settings, relax dtmf and so on, nothing helps. It seems to me, the doorphone's generated DTMF tones are too short. Ok, that I can resolve with some simple hack, but bigger problem is, when secretary presses 8 on the phone, to open the door, doorphone doesn't recognize the tone. Customer has SNOM 190's and BT-100 on their network. Now however long I press the button on the phone, * still sends a very short tone on the line. And that doesn't seem to enough for the doorphone to recognize. Is there any way to make * generate longer DTMF tones? Regards Rennes Neps Cian O'Sullivan wrote: Hello, I have a customer who has their front door integrated to their current phone system. If someone presses the buzzer, the secretaries phone will ring, and she can talk to the person at the door. By pressing ** she can release the door. Anyone have any sort of integration like this. Are there IP devices anyone is using? They have a pizza box server as their asterisk server with a T1 card. No more slots, so if I want to use the existing infrastructure I will need to build a second server with an FXO port. Kinda stupid having a second server just to open the door. Any suggestions? Cian ___ 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] Dial Plan Help
A DigitTimeout(3) will do wonders to (and fix the non existing priorities). Kind regards, E. Versaevel -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens [EMAIL PROTECTED] Verzonden: vrijdag 3 december 2004 21:52 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Dial Plan Help All, I've got a problem here. We are using a Digium 4 T-1 board in our * server. The T-1's are ISDN. The problem I'm having is that we have an ivr setup so that when someone dials our DID it goes to the s extension and starts playing the ivr which is fine, but if someone dials an extension for example extension 200, it doesnt go to 200 it goes to extension 2. Seems like our server doesn't even wait for the rest of the digits dialed. soon as it sees 2 it goes straight to exten 2 and ignores the last two zeros therefore never reaching extension 200. any suggestions? i've enclosed a snipet below. TIA, -Jon exten=s,1,Answer exten=s,2,Wait(1) exten=s,3,Background(intro) exten=s,4,Background(ivrmenu) exten=i,1,Playback(invalid) exten=i,2,Goto(s|4) exten=200,Goto(office,102,1);forward to 102 in office context exten=201,Goto(office,110,1);forward to 110 in office context exten=1,1,Goto(office,102,1) exten=2,1,Goto(office,103,1) exten=3,1,Goto(office,104,1) exten=4,1,Goto(office,105,1) exten=5,1,Goto(office,106,1) exten=0,1,Goto(office,107,1) exten=t,1,Goto(office,108,1) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] SIP SECURITY WARNING: v1-0 (cvs today) sip context in general section ignored goes to default instead - allowing unauthorized sip devices to place calls in default context
Andy Reinke wrote: SIP SECURITY WARNING [general] contex=sip-unauthorized If you spell this right, all calls from unknown SIP devices will be sent to the context you set here. If you do not set a context in the general section of sip.conf, default will be used. This is the way you configure how to receive calls from unknown users, not really a security hole. Everything you define in the [general] context= context will be rechable by anyone. /Olle ___ 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] Users list.
Does this sudden rush of email mean we are all back online? David ___ 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] chan_misdn and Dynalink IS64PH ISDN
Hi, You need to set the layermask when loading the card driver. For a TE port, use 15 (layer 0-3) and for an NT port, use 3 (layer 0-1). Thanks, I add layermask in my modprobe script : /sbin/modprobe --ignore-install w6692pci protocol=2 layermask=3 That would be a TE port with the signaling layer in userspace. For TE, you want the signalling layer in the kernel, i.e. use 15 as the layermask. 3 is for NT ports, because there is no NT signallling in the kernel, so it needs to be done in userspace. Now I have another error : Init. Stack on port 1 TE Stack No lower Id Yep, because there is no signalling layer. I have make a error, if I had layermask in insmod for w6692pci capiinfo don't see anything !! How can I set the layermask parameters ? If you set the layermask to anything below 16, CAPI will not be loaded for that port. chan_misdn conflicts with CAPI, as they both provide the same layer, thus you won't see the chan_misdn ports with capiinfo, this is normal. Simon ___ 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] Voicemail Codec challanges.
Just working on Configing up Voicemail and now that I have got it working and configed and answering the way it should be I have another challange. on the * CLI I get this -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/6001/INBOX/msg format: wav49, 0x8133390 -- x=1, open writing: /var/spool/asterisk/voicemail/default/6001/INBOX/msg format: gsm, 0x8132f48 -- x=2, open writing: /var/spool/asterisk/voicemail/default/6001/INBOX/msg format: wav, 0x8157988 Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space -- Recording automatically stopped after a silence of 10 seconds -- Playing 'auth-thankyou' (language 'en') -- Recording was 0 seconds long but needs to be at least 3 - abandoning -- Playing 'vm-opts' (language 'en') == Spawn extension (default, 8500, 1) exited non-zero on 'SIP/6001-8e4e' when I go to record a voicemail mesg. Anyone got any idea as to which way I would turn? It is likely to be a Config issue but I am unsure were it is to look for it. Thanks for advice in advance. David ___ 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] Recomended ISDN for Asterisk ?
Hi I have installed the http://asteriskathome.sourceforge.net/ with a Digium card with no problems, very good ! Now I want to install my Billion PCI ISDN card (HFC based) in TE mode. I get a little confused with Isdn4Linux, ZapHFC HIAX and the need to install Capi ! Please suggest best and easiest approach ? Thank you ! HB Norway ___ 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] Recommendations for full featured phones
Hi For desk phones I would suggest Grandstream allthough they run at 10m/s so best to seperate the networks Voice and Data. For exec/switchboard extentions go with the Cisco 7960 or Mitel 5220 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sean Cook Sent: Monday, December 06, 2004 1:20 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Recommendations for full featured phones We are considering a replacement of a legacy PBX system (merlin). I am trying to figure out which phones would be best supported with the fullest set of features. Any recommendations? Sean ___ 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 Hardware
Walid Azab [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Can I start using Asterisk with a couple of SIP IP phones and Softphone software on users PCs only? I do not have any cards yet and will still have to wait until I order a card. Yes. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BLOCKING incoming FAXES on voice line.
Joseph wrote: At time to time somebody is trying their luck and send me most likely a junk fax on my voice line. During normal working hours is not a problem I just pickup the line and hangup the call but after-hours my voice mailbox is intercepting the call and recording those beeps (waisting my CPU cycles). Is there a way to block call / issue hangup command if the incoming call is a fax? Assuming you're getting the calls on some sort of a Zap channel, then in the same context where your extention is defined, add: exten = fax,1,Hangup or even better yet: exten = fax,1,Background(if-this-really-is-a-human-please-press-1) exten = fax,2,Hangup You must also have: faxdetect=incoming In zaptel.conf for this to work. Hope this helps, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Is Gigabit Ethernet necessary?
On Sun, 2004-12-05 at 11:57, [EMAIL PROTECTED] wrote: snip [EMAIL PROTECTED] wrote: For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100 Mbps should be fine... The office will have 20 phones, with remote VoIP phones added to the mix later on. If you are using a switch, collisions a pretty much a non-issue, unless you have enough traffic to saturate a port to the server. Latency is also not helped any significant amount, since you still have a 100Mbit link in the path between the phone and Asterisk. Wrong, well, at least it sounds wrong to me. When you look at three concurrent calls between phones and the asterisk server, each phone will have 100MB available between the phone and asterisk (using gigabit). When using 100MB to the server, each phone only has 33MB available. So, with 20 phones, each phone gets 5Mbps to the server, which, bandwidth wise is still plenty, but latency wise, might start to have an impact While, with gigabit, 20 phones can all still have 50Mbps direct to the server In other words, for that application, it likely will not make any difference at all. If it's cheap to do, and the server will also be doing any file serving duties, then it would be a nice insurance policy against a single user swamping the server's port. /snip Well, it probably won't make a huge difference, but I'd probably recommend it, even if just because I don't want the customer to be upset in the future when (if) it does cause a problem ie, overprovision wherever possible... Sounds good to me. The server will be dedicated to Asterisk, so no worries about other applications (unless I move the config to a database which down the line could be very likely). Even then, your bandwidth between the DB and asterisk will likely be quite small... Regards, Adam ___ 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] Door buzzer.
I have the same problem setup with one of our customers, but I have a different problem. I have Grandstream ATA 486 connected to clients doorphone system and clients *. I have two problems: first, when someone is calling from the doorphone, ATA doesn't recognize the called number correctly. 2 or tries out of 10, ATA get's the number wrong. I have tried all kinds of DTMF settings, relax dtmf and so on, nothing helps. It seems to me, the doorphone's generated DTMF tones are too short. Ok, that I can resolve with some simple hack, but bigger problem is, when secretary presses 8 on the phone, to open the door, doorphone doesn't recognize the tone. Customer has SNOM 190's and BT-100 on their network. Now however long I press the button on the phone, * still sends a very short tone on the line. And that doesn't seem to enough for the doorphone to recognize. Is there any way to make * generate longer DTMF tones? Regards Rennes Neps Cian O'Sullivan wrote: Hello, I have a customer who has their front door integrated to their current phone system. If someone presses the buzzer, the secretaries phone will ring, and she can talk to the person at the door. By pressing ** she can release the door. Anyone have any sort of integration like this. Are there IP devices anyone is using? They have a pizza box server as their asterisk server with a T1 card. No more slots, so if I want to use the existing infrastructure I will need to build a second server with an FXO port. Kinda stupid having a second server just to open the door. Any suggestions? Cian ___ 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 + chan_sip2 + sipproxd + sipgate
Am 05.12.2004 um 16:01 schrieb [EMAIL PROTECTED]: Hi, i have an asterisk server behind a masquerading firewall and trying to register to sipgate.de. I use for outbound connections chan_sip2 and on the firewall sipproxd as outound proxy, but it doesnt work. Could anybody help me? register = [sipgate authid]:[sipgate password]:[sipgate [EMAIL PROTECTED]/s ; I dont looked at the rest of the configuration (as I dont know sipproxyd), but this line is wrong. Is has to be : register = [sipgate authid]:[sipgate [EMAIL PROTECTED]/[sipgateauthid] Earlier I thought that the characters behind the '/' are only for defining which extension should ring, but it also get sent to sipgate with the registration and sipgate will complain if this is an 's', it has to be your sipgateid. In you posting you didn't describe what exactly doesn't work, and what exactly you're trying to to, maybe I can help you more if you do. /sebastian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 nativ bridge question?
hallo all, i would like to know, as i would suspect, nativ bridiging should work also, if only one iax party is behind an nat router and the other has a public ip. when i connect to iax clients, which have both pubic ip's nativ bridging is working. if one of the clients is behind an nat, the iax2 channels always get routed through the asterisk server (latest stable version from cvs) ?? i have also set the notransfer=no in iax.conf !! is this normal? in my understanding, it should be possible if one party have a public ip to traverse the udp traffic direct p2p. am i wright? here is what my asterisk server shows during connection (unable to transfer): tahnks, alex Connected to Asterisk CVS-v1-0-12/02/04-14:33:02 currently running on snd (pid = 3792) Verbosity is atleast 5 -- Registered 'atuc' (AUTHENTICATED) at 82.82.238.221:30512 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'snm' logged on from 127.0.0.1 == Manager 'snm' logged off from 127.0.0.1 -- Registered 'streamer' (AUTHENTICATED) at 195.176.254.130:4569 -- Accepting AUTHENTICATED call from 80.141.93.186, requested format = 2, actual format = 2 -- Executing Dial(IAX2/[EMAIL PROTECTED]/3, IAX2/atuc| 10) in new stack -- Called atuc -- Call accepted by 82.82.238.221 (format GSM) -- Format for call is GSM -- IAX2/atuc/9 is ringing -- IAX2/atuc/9 answered IAX2/[EMAIL PROTECTED]/3 -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/3 and IAX2/atuc/9 -- Channel 'IAX2/[EMAIL PROTECTED]/3' unable to transfer snd*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format IAX2/[EMAIL PROTECTED]/380.141.93.186krath 3/2 7/5 [Native Bridged to ID=9] IAX2/atuc/9 82.82.238.221atuc9/21147 5/6 [Native Bridged to ID=3] 2 active IAX channel(s) == Parsing '/etc/asterisk/manager.conf': Found == Manager 'snm' logged on from 127.0.0.1 == Manager 'snm' logged off from 127.0.0.1 snd*CLI iax2 show peers Name/UsernameHost Mask Port Status test/test(Unspecified) (D) 255.255.255.255 0 Unmonitored atuc/atuc82.82.238.221 (D) 255.255.255.255 30512 Unmonitored aleks/aleks (Unspecified) (D) 255.255.255.255 0 Unmonitored hk/hk(Unspecified) (D) 255.255.255.255 0 Unmonitored chris/chris (Unspecified) (D) 255.255.255.255 0 Unmonitored iustus/iustus(Unspecified) (D) 255.255.255.255 0 Unmonitored krath/krath 80.141.93.186 (D) 255.255.255.255 4569 Unmonitored streamer/stream 195.176.254.130 (D) 255.255.255.255 4569 Unmonitored dematuc/dematuc (Unspecified) (D) 255.255.255.255 0 Unmonitored atucek/atucek(Unspecified) (D) 255.255.255.255 0 Unmonitored snd*CLI exit Executing last minute cleanups snd:~# ___ 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] Gossiptel with Asterisk?
Ian Chilton wrote: Has anyone got Gossiptel working with Asterisk? - I am having real problems getting it to register - i'm just getting timeout errors. Yup, I have Asterisk registering with Gossiptel. miranda*CLI sip show peer gossiptel miranda*CLI * Name : gossiptel Secret : Set MD5Secret: Not set Context : from-sip Language : FromUser : 9xx FromDomain : sip.gossiptel.com Callgroup: (0) Pickupgroup : (0) Mailbox : LastMsgsSent : -1 Dynamic : No Expire : -1 Expiry : 900 Insecure : Very Nat : No ACL : No CanReinvite : No PromiscRedir : No DTMFmode : rfc2833 LastMsg : 0 ToHost : sip.gossiptel.com Addr-IP : 193.111.200.14 Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Username : 9307669 Codecs : GSM ULAW ALAW H.263 Status : OK (31 ms) Useragent: Full Contact : ___ 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] Group sip definitions?
Rich Adamson wrote: Been around * for over a year and I'm looking for a way to provide a simple set of group sip.conf phone definitions that can be used by multiple internal sip phones (as in 50/group for example). I have a patch nearly ready that will allow defaults to be specified in any configuration file, that are then treated as if they were entered into each context/category in that file. To make it even more useful, these defaults stack up when you use #include, so you can have customer-specific sip.conf files with their own defaults, and they don't affect any other files you #include into your main sip.conf. If you'd like to help test it out before I post it to Mantis, email me off-line and I'll keep in touch with you; it will be no more than a day or two before it's ready for beta testing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 algorithm?
Hi, do you have info in what countries g.729 is not valid... ? Regards, Robert. - Original Message - From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]; Asterisk Developer Mailing List [EMAIL PROTECTED] Sent: Sunday, December 05, 2004 12:53 PM Subject: [Asterisk-Users] G.729 algorithm? hi all according to what I've found out this far, the G.729 patent seems not valid in a broad range of countries. so... does anyone know where I can find the algorithm? roy ___ 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
