Re: [Asterisk-Users] Cisco 7940 Question
You can have two calls per line appearance. If you assign both line appearances (both can be the same extension) you are allowed four calls. On Mon, 23 Aug 2004, Christopher L. Wade wrote: Hi all, I know this is a stupid question, but it is one I've been trying answer for quite some time. Exactly how many simultaneous calls can the Cisco 7940 have, considering you can be talking to one, and have XXX others on hold? Using SIP, is XXX only 1? I've found documents in various places indicating different values in regard to the max number of calls the phone can handle. I'm just trying to nail down the exact number when the phone is only assigned one directory number (extension). Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jeremy Parr Senior Engineer, Network Services BGC Ltd. ___ 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] Not a User Yet - but have some questions
On Thu, 19 Aug 2004, Peter Harrison wrote: I've been tasked with fitting out an appartment building with a phone system. The entire appartment is getting broadband, with Cat6 Cables into each appartment and fibre down the trunk. I was hoping to have PSTN in the basement connecting to the IP system. Erm, so no POTS to the rooms? What if the user wants to use an answering machine, or a cordless phone? Today I found out however that one of the services will be paid TV - and it will need a PSTN backchannel - ie a modem. The set top devices don't support IP, they have modems built in, and I don't believe they are willing to provide an alternative. Also, the tenants may wish to call IP services via modem, or use a Fax. Why do you need * at all? Why not run the POTS lines directly to the rooms, and cut out all the spinning hard drives, PCI slots, wall wart powered ATAs etc. My question: Can you run modems/Fax over VoIP? Fax/Data in theory should work over ulaw, since there is no compression, but it won't work well if at all in real life. ___ 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] Rodopi Billing
On Fri, 30 Jul 2004, Darren Bentley wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? I suffered with Rodopi for three years in a previous life. Avoid it like the plague. ___ 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] Parking SIP Phones
On Mon, 2 Aug 2004, Jason Williams wrote:
On Sun, 1 Aug 2004 14:50:26 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I use Brian's Valet Parking on our system.
exten = 700,1,ValetParkCall(7${CALLERIDNUM}|mylot|60|${CALLERIDNUM}|1|phones)
exten = _7XX,1,ValetUnParkCall(${EXTEN}|mylot)
To park a call, blind transfer to 700, and to pick it up again, dial
7+your extension. This works well for your small office.
This will only work if you have two digit extensions
Jason
exten = 7000,1,ValetParkCall(7${CALLERIDNUM}|mylot|60|${CALLERIDNUM}|1|phones)
exten = _7XXX,1,ValetUnParkCall(${EXTEN}|mylot)
And now it magically works with three digit extensions. Do you need me to
paste the config for four digit extensions as well?
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RE: [Asterisk-Users] 480i User Feedback With Asterisk 802.1Q?
On Sat, 31 Jul 2004, Kevin wrote: Does anyone know if the 480i supports 802.1Q? I don't see any support for it at the moment, but this is a very early firmware, with a bare minimum of features. ___ 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] Parking SIP Phones
On Sun, 1 Aug 2004, Trevor Peirce wrote:
Hello,
I know not too long ago I saw /something/ _somewhere_ about an
adjustment to call parking that allowed blind transfers from SIP phones
to park a call and still be able to hear the parking lot stall number.
Unfortunately, I have no idea where I saw that (google turned up little,
couldn't find it on the list either). I'm using Sipura SPA-2000
adapters and it doesn't seem to work with today's CVS.
I use Brian's Valet Parking on our system.
exten = 700,1,ValetParkCall(7${CALLERIDNUM}|mylot|60|${CALLERIDNUM}|1|phones)
exten = _7XX,1,ValetUnParkCall(${EXTEN}|mylot)
To park a call, blind transfer to 700, and to pick it up again, dial
7+your extension. This works well for your small office.
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[Asterisk-Users] Message Lamps across IAX connected switches.
I have an * server at my office, lighting message lamps on SIP phones fine. I also keep an * server at home, with no Zaptel hardware, and hang a SIP phone off of it so I have an extension at home with no SIP/NAT ugliness. Is there a way to light the message lamp at home when I have voicemail at my office? ___ 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] 480i User Feedback With Asterisk (fwd)
For those that are interested, here is my report back to Sayson on the 480i -- Forwarded message -- Date: Sat, 31 Jul 2004 22:03:31 -0400 (EDT) From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: 480i User Feedback With Asterisk Seshu, I am using a 480i, and I am impressed with the phone on a whole, but obviously the firmware is lacking. Details follow. Hold button works, but holds the user at the phone, does not hold them at the PBX, allowing for music on hold. I would also like to see the hold button software addressable so that it could be used to park the call (transfer to the parking extension) rather than only putting the call on hold. Transfer seems to work fine, but would like to see blind transfer (transfer direct to the remote party, rather than the current behavior where you are bridged to the called party, then have to press transfer again to complete the transfer) Maybe a softbutton for blind transfer, and reserve the hard button for attended transfers? It would be nice if we could change the behavior of the transfer button to do this. Conference works, but I have some trouble dropping the correct party from conversation. Also, when conferenced, I am no longer able to send DTMF tones to the second line. The original caller hears the DTMF, but the second party does not. I am using RFC2833 DTMF. Redial seems to work a bit odd. It does redial a previous number, but not the last number dialed. Can't seem to find any rhyme or reason, but it seems do dial the last non PSTN number. For instance, if I dial 50 for voicemail, then dial 93934481, hang up, and press redial, it dials 50, not 93934481. When viewing the SIP config through the menu on the phone, it displays defaults, and not the settings specified via the TFTP configs. If I set qualify=1000 in my sip.conf, Asterisk will send a poke to the phone every few seconds, to make sure it is still alive. If I enable this option, the phone stops working after a few minutes. Asterisk shows the phone as unreachable, and I cannot dial any number from the phone. It will accept the input, but the dialplan does not timeout and dial, nor will it dial if I press the # sign. On the subject of dialplans, I am only able to dial 10 digits, on the 11th digit, the phone tries to dial. This is a bad thing when trying to dial long distance. It is basically impossible. The display occasionally shows L1 in the lower left hand corner of the display. As if someone had pressed the left/right arrows of the navigation pad. It shows up for less than a second, and then goes away. On to the good things now. ;-) Conference basically works, save for the caviats above. The backlight is very nice, and the surface of the LCD does not show fingerprints like the Cisco phones do. Sound quality is good with ULAW codec, but there is noticable echo in the call that is not present when using a Cisco phone, or a Softphone. Transfer seems to work fine. Message lamp and stutter dialtone works fine. I would like to have a way to turn off the stutter tone. If I have a message lamp, there is no need for a stutter tone. The web config is nice, but limited. I would like to be able to set up SIP configs through there. Also it only works from Internet Explorer on Windows. If you are running Netscape/Mozilla/Opera, you cannot use the web config. It should also be fully password protected, so rouge users cannot reboot the phone etc. Custom ringers are nice, and professional, recognize them all from Nortel phones. I would like to be able to specify a SIP Alert_Info so that I can have a different ring from different calling partys. For instance, incoming calls from the PSTN would have one ring, and extension to extension calls would have another ring. Look forward to hearing back from you. Nice work on this phone. If it hits the market with a complete firmware, at $200 or less, they will sell well within the Asterisk community. Jeremy Parr Senior Engineer, Network Services BGC Ltd. ___ 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] DTMF stops working w/ Voicemail
On Fri, 23 Jul 2004, Brent Franks wrote: Hello, I have some reports from users that occasionally DTMF will stop working in voicemail and they will have to exit the system to get it to work again. The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with Ulaw codec. This is all on an internal switched 100mb lan. Has anyone else seen anything like this? I have noticed it before, but haven't seen any way to reproduce the problem. Using Cisco phones with OOB dtmf. ___ 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] uip200 clips audio prompts
On Mon, 19 Jul 2004, Ryan Courtnage wrote: Hi all, We find that our UIP200s clip off the 1st second of audio prompts from * (ie: the beginning of voicemail prompts). Has anyone found a way around this? Running CVS-D2004.06.29.15.30.00 on WBEL 3.0 This happens with my 7940s as well. I have found that using and Answer, and a Wait(1) before playing back prompts works well. Prevents Alisson from saying Assword? when dialing VoicemailMail(20). ___ 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] uip200 clips audio prompts
On Mon, 19 Jul 2004, Ryan Courtnage wrote: This happens with my 7940s as well. I have found that using and Answer, and a Wait(1) before playing back prompts works well. Prevents Alisson from saying Assword? when dialing VoicemailMail(20). Thanks for your reply. I have been able to use this method to eliminate some of the problems, but from within the voicemail application, I don't beleive there is a way to set a delay between each prompt? ie: I'll hear: Press 0 for New messages, ... for old messages, ... for work message . The Press x.. is cut off of the beginning of the prompts. I only see this problem with uip200s. BT102s, handytones, sipuras, etc work just fine. Could it be silence supression? ___ 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] LAN Switch w/ QoS
On Sun, 18 Jul 2004, Michael Welter wrote: Does anyone have a recommendation for a 48 port LAN switch for a new * system? I'm not happy with NetGear's reliability. You can get Cisco 2950s for about $600/24 ports. ___ 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] NuFone Error
You need to send a vallid CALLERID to Nufone. On Sat, 10 Jul 2004, V59Net wrote: Use the NuFone to call numbers 1800,1866 and after 20 seconds the call is interrupted. In log of * it writes: Max retries exceeded you host. Somebody can help me? ___ 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] NuFone Error
On Sat, 10 Jul 2004, Brian K. West wrote: No you don't it will just make one up... I beg to differ, Mr Brian sir. I had problems calling 800 numbers with Nufone, and Jeremy explained to me that they check for a caller id before sending calls to 800 numbers. ___ 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] IRC channel #asterisk on irc.freenode.net
On Fri, 9 Jul 2004, Antti Lohikoski wrote: and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) Maybe your username is invalid. ___ 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] IRC channel #asterisk on irc.freenode.net
On Fri, 9 Jul 2004, Andrew Kohlsmith wrote: On Friday 09 July 2004 06:54, Antti Lohikoski wrote: 1. The irc.freenode.net server gives me Couldn't look up your hostname and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) This is *specifically* why I wish bkw (Brian West) would turn off that flag on the channel. In order to combat spam bots infiltrating the channel, it is set up to only allow freenode-registered nicknames. In order to register your nickname with freenode, send a /msg nickserv help command once you're on freenode. NickServ is a Nickname Server bot -- it will let you register a nickname and set a password so your nickname can't be stolen. Identd is *not* required. Ok guys, enough FUD and wrong answers. He cannot get on, because his USERNAME has invalid characters antti.loh is not valid. You cannot have . in your username. This is NOTHING to do with the channel requiring you to register with nickserv, this is NOTHING to do with ident. Antti, in your IRC client, you are given a choice of nickname, and realname/username. Make sure both of these are a-z/0-9, no special characters. You should be fine. ___ 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 static - out of ideas
On Wed, 7 Jul 2004, Ryan Courtnage wrote: Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. Attempts to send outbound calls on any Zap channel will result in hearing a loud 'static' noise on the line. On one occasion the problem actually occurred while someone was on an active call with the PSTN. 25 minutes into the call, this loud static noise occurred, and the call was dropped. Debug log files show nothing unusual. It's obvious that * is unaware that there is any problem with the card. I had a simmilar problem with an FXS card in a Compaq ML330. I would get a power reset message on my server console. Are you sure the noise is coming from the FXO ports? Are you using SIP phones, or ZAP? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re:Latest Echo changes
On Wed, 30 Jun 2004, Brian McSpadden wrote: What settings are you using for rxgain/txgain? Do you have echotraining=yes on? On Mon, 28 Jun 2004 00:45:52 -0500, Chris Foster [EMAIL PROTECTED] wrote: I Just finished getting HEAD running (had to update to Slackware 9.1 from 9.0 to do it) and much of my X100P's echo is gone. This is without having to change any settings except those which we're way off (my rxgain/txgain was wildly off). I don't even have echocancel=800 on. the 800 is echotraining=800. ___ 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] TDM411B configuration
On Tue, 29 Jun 2004, rich allen wrote: iH using Slackware 9.1 after install the card (new) i get the following from dmesg Module 0: Not installed Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) how can i determine if this is a hardware failure or an IRQ issue You aren't running the latest Zaptel from CVS, are you? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re:Latest Echo changes
On Sun, 27 Jun 2004, taf taffey wrote: Hi, I've tried the latest CVS Zaptel and Asterisk after following the Echo fix threads. But echo is the same if not worst. Has anyone managed to alleviate their echo from these latest changes? My echo has vanished entirely with the latest CVS. TDM04B, echotraining=800 ___ 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] Don't want a ring before voice menu
On Tue, 8 Jun 2004, John Campbell wrote: Hi, Having searched through the mailing list archives and the wiki, I still don't know how to solve the following problem: Call is received, phone rings once, then the caller gets the voice menu. What I want is for the call not to actually ring, but to go straight to the voice menu. How can I achieve this? Thanks, You are using analog lines? If so, asterisk has no way of knowing the phone is ringing, until it rings. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: Voicemail and Cisco Phones
On Mon, 7 Jun 2004, Kurt wrote: The Cisco 7960 has a softkey called DND which when pressed as the phone is ringing will sack the call to voicemail. If you where using Cisco CME or CM you can forward all calls to Vmail via CLI or GUI. It does? I have 7940s, and the DND is buried deep inside some menus. ___ 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] AM-Web working?
I am trying to run am-web on my asterisk server. The machine is CVS-HEAD
from 5-29-2004, on Debian Testing, running Apache as httpd.
If I untar the am-web.tar.gz file to /var/www/am-web, and access
http://office.bgcfreedom.com/am-web//command.php?page=listsip or any other
command in a browser, it returns this error:
Warning: main(): Failed opening '' for inclusion
(include_path='.:/usr/share/php:/usr/share/pear') in
/var/www/am-web/command.php on line 23
Examining command.php, I see this:
3 $include = ;
22 {
23 include($include);
24 }
If I remove those lines, the include error is gone, but now every command
page I load returns:
htmlbody/body/html
There appears to be a CVS version of the software, but the server cannot
be resolved. Could someone with a working am-web install mind posting
their command.php?
Thanks
Jeremy
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Re: [Asterisk-Users] Stuck SIP channels? - SIP show channels
I see the same thing: marconi*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format 192.168.1.100(None) 000f9054-3a 00101/02439 UNKN 192.168.1.103(None) 000f9057-96 00101/01079 UNKN 192.168.1.101(None) 000f9048-5d 00101/00113 UNKN 192.168.1.101(None) 000f9048-5d 00101/00113 UNKN 192.168.1.103(None) 000f9057-96 00101/01079 UNKN 5 active SIP channel(s) marconi*CLI show version Asterisk CVS-HEAD-05/29/04-15:18:19 built by [EMAIL PROTECTED] on a i686 running Linux marconi*CLI Is this something to worry about? Or Normal behavior? On Tue, 1 Jun 2004, Karl Brose wrote: Yes, it is something to worry about, because you might run out of RTP ports or open fd's, depending on your port range in rtp.conf. Which cvs version are you running? This behavior was observed by several people for a short period of time and then seemed to have disappeared with a cvs versions starting around 1.390 - 1.394 (chan_sip.c) according to my observations (more like a guess actually) couldn't exactly pinpoint the patch that stopped it. Mickey Binder wrote: Hello all I've discovered that SIP channels sometimes get stuck in *. I've read some posts from Fri 29 Aug 2003 which mentions this issue, but there doesn't seem to be any final answers I don't know if this is related to the 0001604 bug? Below is a list from one of the incidents: I know the (d) means that it is scheduled for destruction but the 10.1.1.45 channel hasn't been used for a couple of days. My setup consists of two different brands of devices. The Peer with IP 10.204.10.12 is an AudioCodes MP-124, and every other IP is a number of Welltech 3504A 4-port FXS devices. asterisk-srv1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format 10.204.10.12 1619132d381f58106 00103/0 UNKN (d) 10.204.10.12 1619131e68f3610c9 00103/0 UNKN (d) 10.204.10.12 46391328862156821 00102/05918 UNKN (d) 10.204.10.12 46894525028137781 00102/16213 UNKN (d) 10.204.10.20 30512957f9ac-acc0 00102/2 UNKN (d) 10.204.10.15 46704057f9cc-acc0 00102/2 UNKN (d) 10.204.10.15 4670405800ec-acc0 00102/2 UNKN (d) 10.1.1.459096172f560a1e6dc 00103/0 UNKN (d) 10.204.10.14 83208257fd3c-acc0 00102/2 UNKN (d) 10.204.10.12 9096174f8157c61b3 00103/0 UNKN (d) 10.204.10.12 90961758022c-acc0 00102/2 UNKN (d) 10.204.10.12 9096172b2763885b7 00103/0 UNKN (d) 10.204.10.23 44801957faec-acc0 00102/2 UNKN (d) 10.204.10.19 30302057ffcc-acc0 00101/4 UNKN (d) 10.204.10.13 45871757fc2c-acc0 00102/2 UNKN (d) 10.204.10.13 45871757fafc-acc0 00101/2 UNKN (d) 10.204.10.13 45871757f89c-acc0 00102/2 UNKN (d) 10.204.10.15 46427758034c-acc0 00102/2 UNKN (d) 10.204.10.15 46704057fafc-acc0 00101/3 UNKN (d) 10.204.10.15 46704057f89c-acc0 00102/3 UNKN (d) 10.204.10.24 9096675805ac-acc0 00102/2 UNKN (d) 10.204.10.15 46704057fe8c-acc0 00102/3 UNKN (d) 10.204.10.19 90965758035c-acc0 00102/2 UNKN (d) 10.204.10.13 45871757faec-acc0 00102/2 UNKN (d) 10.204.10.19 90965758022c-acc0 00102/2 UNKN (d) 10.204.10.15 46566858021c-acc0 00102/2 UNKN (d) 10.204.10.14 46894557fadc-acc0 00102/2 UNKN (d) 10.204.10.14 4689455800cc-acc0 00102/2 UNKN (d) 10.204.10.22 90965357f9dc-acc0 00102/2 UNKN (d) 10.204.10.22 90965358048c-acc0 00102/2 UNKN (d) 10.204.10.15 46427757ffbc-acc0 00102/2 UNKN (d) 10.204.10.14 46894557fd3c-acc0 00102/2 UNKN (d) 10.204.10.14 46894557fe6c-acc0 00101/3 UNKN (d) 10.204.10.14 46894557fadc-acc0 00102/2 UNKN (d) 10.204.10.15 46427757ffbc-acc0 00102/2 UNKN (d) 10.204.10.19 30302057fc3c-acc0 00102/2 UNKN (d) 10.204.10.17 17849957fd3c-acc0 00101/3 UNKN (d) 10.204.10.18 44281157ffbc-acc0 00102/3 UNKN (d) 10.204.10.13 45871757f9bc-acc0 00102/2 UNKN (d) 10.204.10.20 30512958032c-acc0 00101/3 UNKN (d) 10.204.10.19 30302057f9dc-acc0 00102/2 UNKN (d) 36 active SIP channel(s) Is this something that I should worry about? regards, Mickey Binder ___ 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]
Re: [Asterisk-Users] tdm04b stopped taking inbound calls - todays cvs: CONFIRMED
On Wed, 26 May 2004, Rich Adamson wrote: Has anyone tried the current Head cvs with TDM04b (4-port fxo)? The card stopped answering inbound calls (no CLI indications whatsoever), although outbound pstn calls via the card work just fine. Kind of looks like one of the changes from yesterday (probably wcfxs.c) might be causing the problem. (Total new checkout, install, reboot, etc, result in the same no-answer condition.) Anyone else seeing this? Yup, I had the same problem, updated from CVS last night. I was able to check out 1 day old zaptel code, and it worked fine. TDM04B ___ 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] VoicePulse SIP
Welcome to Voicepulse and their lack of jitter buffer. This is the cause of your horrible sound. Will be just as bad with SIP. Which providers give you a jitter buffer? ___ 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] Power alarm on module 1, resetting.
Any clues as to what this is? Google isn't much help. Both cards have the power connector plugged in, and both randomly give me static instead of a dialtone, or an outside line. Cards are 1 4 Port TDM400 FXO, and 1 4 Port TDM400 FXS. Computer is a Compaq Proliant ML330 The full error is as follows: Ouch, part reset, quickly restoring reality (0) Power alarm on module 1, resetting! ___ 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] Dead FXO Module on TDM400P?
Since the irc channel wasn't any help, I will try posting my problem here. I have two TDM400Ps less than a week old in a PC. All of the FXS ports work fine, and all of the FXO ports worked fine up until thisafternoon. If I try to dial in, I get a busy signal, if I try to dial out, all I hear is a very scratchy, very crackly dialtone. If I swap the first FXO module with the second on the card, the problem follows the module, not the physical port on the TDM400P. Any thoughts? Was planning on deploying this server tomorrow, but now I am worried about the other three FXO ports crapping out. Has anyone else had reliability problems with the new FXO card? Or is it too early to tell? ___ 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
