[Asterisk-Users] Switch brands, speeds, etc.
The short of it: In light of the recent Netgear posts, I'm just curious if anyone has preferences for brands of switches - we're wiring a parallel network of 10BaseT over existing cat3 for the IP phones in our office space. The long of it: --- Our setup: * Office of 10 people spread out in 2000-3000 sq.ft. * Space previously used as computer learning center, chock full of cat-3 and multiple rj-45 jacks per wall plate. * We're rewiring anyway - company growth + lack of planning has led to switches and hubs strung everywhere * I've convinced the boss to let me implement an asterisk server, replacing the unholy phone concoction we have now * No external VOIPat least not yet. * MUCH data flying back and forth from computers in labs to offices and vice versa So we were thinking of using some of the existing cat3 for just the IP phones and stringing some cat5e alongside for intranet. Buy a cheap 10BaseT switch (SmallDog has a refurbished Asante 5324 24-port cheap) for the cat3 lines and feed that to our * server's eth1. We're geeks, but not really networking geeks, so I thought I'd ask the list populace at large if they had comments/recommendations. Best, Bob Klepfer Photon-X, Inc. ___ 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] billing question
On Thu, 2004-02-12 at 09:52, Arretni VoIP Tech wrote: hello, Is it normal that * starts its billing when voicemail starts to prompt? can I do something like it will only start to bill if the caller left a message? right now, im seeing that unanswered calls that are forwared to voicemail are considered billable as well as calls to voicemailmain. Since the PSTN bills based on connect time, this is what Asterisk is doing. The circuit is busy as soon as you answer the line regardless of the action the user has. -- Steven Critchfield [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
[Asterisk-Users] Wierd Zap Channel Behavior
Here's a wierd one. I'm have a problem where periodically a couple of my extensions dont' get hungup properly. The channel bank doesn't show the channel as active, show channels doesn't show the channel as active, but a zap show channel has the Actual Confinfo: as an active call. This results in the channel receiving one-way audio from an active conversation on another Zap channel. I'm running: Zaptel CVS 2-10-04 (for bigzaplock fix) libpri 0.5.1 asterisk 0.7.1 This happened with zaptel 0.8.1 as well. My guess is that asterisk isn't properly closing the channel when it's hungup. Has anyone seen this behavior? Here's the output of zap show channel 37 File Descriptor: 111 Span: 2 Extension: Context: longdistance Caller ID string: Bad Extension 5239 Destroy: 0 Signalling Type: FXO Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/161, Mode/0x0009 Actual Confmute: No ___ 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] FWD/Iaxtel/Asterisk codec use
The codec issues with different services and sip phone are the most complicated and trusting experience when using Voip services. I had been able to connect to FWD behind a firewall by using Iaxtel using g729. Just recently, about a week, every time I tried to call FWD, the connection simply timed out. The console message says the circuit is busy. Or every one is busy at the moment. However, when I change the codec of the connect to GSM. The connection is back to normal. It may be due to changes to Iaxtel or their G729 licence runs out of capacity. When I fiddled the iax.conf file, in the [iaxtel] section, I specified disallow=all allow=gsm disallow=g729 it still does not work. then I have it change the [general] section as well disallow=all allow=gsm disallow=g729 It then works. Is there any logical explanation to this. I wonder. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] multiple context in sip.conf
Hi, Is it possible to have multiple context= for user configuration in sip.conf? Regards, Antonio Rabena ___ 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] [OT] Looking for Manual: Clarent CPG 101
Hi, My client acquired a Clarent CPG 101. No included manuals except for the startup guide. Would someone have an electronic/PDF manual? I can't even find the product name on the web. Thanks in advance. ___ 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] AudioCodes MP-104 - help !
I have really got things upto a point where I have no clue why Asterisk doesnt authorise the audiocodes fxs box. I can not find anything in the archives - some posts that could actually be useful are already deleted. Can anyone please help me out ?! My setup: PSTN - Asterisk ---router/nat-- Audiocodes - POTS xxx.xxx.xxx.xxx 80.54.223.79 192.168.0.1192.168.0.249 Debug Log: Sip read: REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKacEKUOEAs From: sip:[EMAIL PROTECTED];tag=1c20095 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 16212 REGISTER Expires: 3600 Contact: sip:[EMAIL PROTECTED];user=phone;expires=3600 Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 80.54.223.79 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKacEKUOEAs;received=80.54.223.79 From: sip:[EMAIL PROTECTED];tag=1c20095 To: sip:[EMAIL PROTECTED];tag=as0b288949 Call-ID: [EMAIL PROTECTED] CSeq: 16212 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 80.54.223.79:1025 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKacEKUOEAs;received=80.54.223.79 From: sip:[EMAIL PROTECTED];tag=1c20095 To: sip:[EMAIL PROTECTED];tag=as0b288949 Call-ID: [EMAIL PROTECTED] CSeq: 16212 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=3465f8ca Content-Length: 0 to 80.54.223.79:1025 asterisk*CLI Sip read: REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKactrVcPNj From: sip:[EMAIL PROTECTED];tag=1c20095 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 16213 REGISTER Contact: sip:[EMAIL PROTECTED];user=phone;expires=3600 Proxy-Authorization:Digest username=mp_104_test,realm=asterisk,nonce=3465f8ca,uri=sip:xxx.xxx.xx x.xxx,Algorithm=MD5,response=a41319cc5c8f9ddab6be04b2afe3d0ba Supported: em,timer,100rel Expires: 3600 User-Agent: Audiocodes-Sip-Gateway/MP-104 FXS/v.4.20.299.410 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 80.54.223.79 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKactrVcPNj;received=80.54.223.79 From: sip:[EMAIL PROTECTED];tag=1c20095 To: sip:[EMAIL PROTECTED];tag=as0b288949 Call-ID: [EMAIL PROTECTED] CSeq: 16213 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 80.54.223.79:1025 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKactrVcPNj;received=80.54.223.79 From: sip:[EMAIL PROTECTED];tag=1c20095 To: sip:[EMAIL PROTECTED];tag=as0b288949 Call-ID: [EMAIL PROTECTED] CSeq: 16213 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 80.54.223.79:1025 Feb 14 12:39:49 NOTICE[1133718080]: chan_sip.c:5405 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '80.54.223.79' in sip.conf [mp_104_test] type=friend username=mp_104_test secret=mp_104_test auth=md5 disallow=all allow=g729 allow=alaw host=dynamic nat=yes qualify=200 dtmftone=rfc2833 context=default On the AudioCodes gateway I use authentication per endpoint. Thanks - I would really appreciate any help what so ever Dave ___ 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] CAPI noise and scratches - strange behavior
Hi all, Here are some additional info findings: My system is dual PIII 933MHz, with a Gigabyte GA-6VXDC7 mobo. Asterisk produces noises and scratches when the system is IDLE and *BOTH* CAPI channels are active (with one CAPI channel everything is fine always). When the system is busy (CPUs at 100%), sound is perfect with *BOTH* CAPI channels active. Here is how I can reproduce the problem: If I keep one of my CPUs busy while both CAPI channels are active, the noises stop. Just a scratch here and there. I keep the CPU busy with this: # while [ 1 = 1 ]; do echo /dev/null; done If I run this twice (to keep busy both CPUs), then the noises appear just every a few seconds. I can reproduce this every single time. Note that I have a TDM40B (4xFXS) Digium card installed that is supposed to provide a high resolution timer to asterisk. The effect is the same with this card enabled and disabled. I know the easy workaround: Run setiathome permanently :-) However, it seems that chan_capi has some timing problems... Any help? Costa On , 2004-02-11 at 14:08, Costa Tsaousis wrote: Hi, When I have two concurrent CAPI calls, * produces a lot of noises and scratches on both CAPI channels. I am using SIP phones; it appears on all phones, even if two separate SIP devices are connected to the two CAPI channels. The problem does not appear with any number of concurrent calls using SIP end-to-end (with * as a media gateway). I am using FritzCard! DSL (the ISDN part of it) and kernel 2.4.24 vanilla. Any help is appreciated. Costa ___ 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 problem with IpDialog phone.
Turn sip debug on and forward the logs. A 481 means that a dialog was not correctly established. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ariel Batista Sent: Thursday, February 12, 2004 6:28 PM To: Asterisk User List Subject: [Asterisk-Users] Sip problem with IpDialog phone. I have one of my IpDialog phones giving this error about once an hour. On the Asterisk server CLI I get this message. Got SIP response 481 Call Leg/Transaction Does Not Exist back from 204.241.XXX.XXX If I go to the phone and dial out it works and I no longer get the message. Also if I check the sip show channels I get 2 additional connections with unknown information for the IpDialog phone. Other then this message the phone work fine. But when the message comes up I can not dial call the phone. ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] Problem with * - Nat - Internet - Nat - X-lite
I have an * server installed inside a NAT that is configured to DMZ the address of the server. This means that all traffic is automatically redirected to *. When I try to connect from a Windows machine (using X-lite) that is behind another NAT I connect and looking at the * console I can see that X-Lite is registering. The problem is that audio seems to drop out. When I dial another extension it rings and the other person can answer the call, but no audio is heard on either side. If I dial the Voicemail menu I can hear the prompt for the password and then it starts to list the number of voicemails in my mailbox but audio cuts out. By watching the console I can see that everything is working, the voice prompts are being sent and I can press the number for an option. Every time I press a number I do get the first prompt and then audio goes out again. As far as I can see everything is working correctly. The same notebook computer works perfectly when on the same network. Here is a little snip of my sip.conf [general] port = 5060 bindaddr = 0.0.0.0 externip = (real Internet address) localnet = 192.168.0.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = (my context) ; Default for incoming calls [4002] type=friend username=4002 secret=secret host=dynamic amaflags=default accountcode=temp callerid=Carlos Chavez 4002 nat=yes mailbox=4002 The other extensions I am trying to call are in the same internal network as the * server. Any suggestions? -- Carlos Chavez Computer Engineer, CCNA Corporativo Lacer S.A. de C.V. ___ 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] Digium connectivity issue?
Rich Adamson wrote: Are others seeing hugh delays and/or lack of connectivity to Digium? 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 I saw an over 12hr outage to Digium and IAXtel and I know it wasn't OUR network so, they must have had some sort of problem. John [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] Call Queues
On Tue, 10 Feb 2004, Jonathan Stanton @ Home wrote: Any ideas / sugestions welcome. Having the queue calls delivered to an agent login would appear to be the easiest way to do it - just log in the agent on any phone you like, and calls will be diverted to that phone. As another poster noted, one little gotcha is that if your announcement file is missing, the calls will be disconnected when you try to answer them. -- Jon Stockill [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] Digium connectivity issue?
I am having same problem and i was never successful in connecting to digium.com or asterisk.org or asteriskpbx.org for last three days. Deepak - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk-a-users-list [EMAIL PROTECTED] Sent: Friday, February 13, 2004 01:54 PM Subject: [Asterisk-Users] Digium connectivity issue? Are others seeing hugh delays and/or lack of connectivity to Digium? 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 ___ 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] festival voices
At 2:56 PM -0800 2/12/04, Chris Albertson wrote: --- Tony Buser [EMAIL PROTECTED] wrote: Hi, I'm new to both asterisk and festival. I'm trying to figure out how to change the voice festival uses. For example, I've downloaded don_diphone to festival/lib/voices/english. I then edited /etc/asterisk/festival.conf and changed the festival command to: festivalcommand=(voice_don_diphone)(tts_textasterisk %s 'file)(quit)\n try adding a set of parens like this: festivalcommand=((voice_don_diphone)(tts_textasterisk %s'file)(quit))\n SNIP natural sounding voice? So far the best I've found were from here: http://hts.ics.nitech.ac.jp/download.html Have you seen festivox? It's a tool for building voices The key to making festival sound natural is to get the timming and entonation right. The astrisk app uses festivels demo test to speech application which is just that a quick dirty demo. Have you seen the markup language on the CMU site? http://www-2.cs.cmu.edu/~awb/festival_demos/sable.html Sable can do MUCH better then the simple tts application. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK As a reminder to our viewing audience: search the archives for cepstral - there are some decent sounding voices with Cepstral, for $30. There is a patch in the bugtracker for app_cepstral, though I have not had a chance to play with it yet. JT ___ 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] GSM codec with Cisco equipment
I had been under the impression (after actual tests) that the GSM codec supplied with Asterisk was not compatible with the GSM codec supported by the Cisco VoIP equipment (PRI and DS3 gateways.) However, I hear recent news that this is no longer the case. Can someone with a big Cisco media gateway (58xx or similar) please test and let the community know if this works with recent IOS images? I will attempt to test with a small media gateway in the near future, but my schedule is pretty much booked solid for the next week. JT ___ 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] Anybody going to the Spring VON converence [ OT]
On Thursday 12 February 2004 18:15, Bob Knight wrote: Not sure if I will attend VON, but myself and a friend would be way into an * nerd fest. http://www.interz0ne.com/ http://www.phreaknic.info/ Mark Spencer and a few other Asterisk developers are likely to attend both. Note that while Phreaknic 8 will take place, the date won't be set until the Titans' football schedule is released, as the hotel is located close to the stadium. -Tilghman ___ 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 / Echo / ZTMONITOR CAN2,3, etc.
On Thursday 12 February 2004 19:48, Jeff Stohl wrote: My echo is really bad on the dialing out line only, the other user has a perfect reception. The inside user has just a horrible echo back of their voice. I've tried various echo canel methods in zapata.conf as well as regular TX/RX gains. I tried running ztmonitor 1 -v and the ztmonitor RX and TX charts came up but did not provide any display back so I was unable to find some better TX/RX. I tried several settings but those didn't help. If you're getting echo of your own voice, but the remote is getting a clear signal, then Asterisk echo cancellation is working properly. It is the remote provider not echo cancelling properly. Think of the echo in this way: what you're hearing is your voice on the remote receiver being picked up by the remote microphone and sent back to you. You might ask your remote party not to use a speakerphone and to press the receiver tightly against his/her ear, as this will dampen the sound detected by the microphone and lessen the echo. -Tilghman ___ 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] Anybody going to the Spring VON converence [OT]
I was wondering if anybody was planning on attending Jeff Pulver's Spring VON conference in Santa Clara. I am thinking about going and was hoping, if enough Asterisk people are going to be there, if we couldn't hold some kind of ad hoc Astericon or something. Perhaps we could rent a room at the hotel for an evening get together or something? I know Mark will be there. Might be kind of fun. Also, I am still trying to decide if I should go for the full conference or just a few days. Any thoughts? Steve I suspect I will be there, in some disguise or other. This is a good segue into the beer and chat session we had last night here at the Internet Telephony Expo in Miami. We had about 11 people show up, and good chatter ensued. Attendees included representatives from several local Miami firms that use Asterisk, the folks from Snom (and distributors), and some various riff-raff such as myself. Thanks to Alex Lopez for the cruise directing to the ritzy rooftop pool/lounge/art studio place overlooking Miami's skyline. I extracted information from the Snom people that was very interesting: they have a large-screen deskset phone which is coming out shortly. It's the 220, and it also has a programmable (SIP!!) module that provides up to 65 keys. And, perhaps the coolest gee-whiz feature is that it supports the Alert-Info: header with a URL that points to a sound file, so you can send custom ring tones with EACH CALL, perhaps even doing pre-announce. Cool. JT ___ 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: Hide outgoing CallerId on Zap interface
Mickey == Mickey Binder [EMAIL PROTECTED] writes: Mickey I want to completely hide my outgoing CallerId when dialing Mickey out on my Zap interface. What kind of zap interface? If it is an fxo card on a standard pots line, treat it as such and prefix the dialed number with the right incantation. I don't think there is a wait-for-tone letter -- unlike in a modem's ATDT command the W is used for pause on the zaps -- so you'll need to experiment to find how many pauses you need. If it is a pri I'd give SetCallerID() a try in the dialplan. For a non-pri t1/e1 perhaps someone else can shed some light? -JimC ___ 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] Easy access to visual busy status and call transfer buttons
I want to say thanks for the great posts to this list...I learn something know about every day reading this list. Anyway...I have been using * in a test environment for 10 months and really like it. I have PRI to the PSTN and SIP to 2 Snoms and 1 Cisco 7960. I have frequently used ATT/Lucent/Avaya phone systems such as Definity or Partner that provide the ability to assign LEDs on individual phones that allow you to visually see the status of specific extensions to determine if the extension is on a call, do not disturb, or idle. If I use * to speak SIP to the phones...such as the Cisco 7960...how do you provide users with this easy visual way to see the status of an extension? Further...using a button associated with these busy status indicators makes transferring calls fast. I see some people use software on a PC to get this functionality. It still seems that there should be a way to do this on a SIP phone. Am I the only person that thinks these status LEDs are valuable? Jeff ___ 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] Digium connectivity issue?
On Fri, Feb 13, 2004 at 07:54:50AM -0600, Rich Adamson wrote: Are others seeing hugh delays and/or lack of connectivity to Digium? Yeah. Traceroutes stop at their provider. They are either having connectivity issues (if they have dsl/T1/whatever into their facility) or server problems if colo-ed. Digium and * websites are down at the same time... Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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!!!! Having problems Starting Asterisk
On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote: I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have re installed mgp123 but it still doesn't help any Ideas? Try shutting down all * processes (including mpg123). Now, see if your audio works normally. If not, rmmod the zaptel/fx? modules, and see if that works. If not, you should start by getting your audio on the consloe to work normally first, then, check with the zap/etc modules loaded, then try * . One step at a time. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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: Rhino channel bank and aastra PT390 phones
Hi, Anyone setup a Rhino channel bank ?... any issues. I got it working with normal pots phones but I cant get it to work with Aastra PT390 phones. The phones get dialtone but the asterisk does see any DTMF digits dialed from the phone. Any ideas would be helpfull. Thanks John Bittner Simlab.net ___ 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] Jitter Buffer Configuration (typo in iax.conf)
Hi I havent been able to get the jitter buffer to work even with correct typing. If you have any luck, please let me know how it performs for you. Thanks and Regards Clive On Thu, 12 Feb 2004 19:56:27 + (GMT) Michael T Farnworth [EMAIL PROTECTED] wrote: I had noticed that the jitterbuffer settings under Asterisk didn't seem to work very well, then I noticed that there was a typo in my iax.conf file where I had: maxexccessbuffer=750 which should have been maxexcessbuffer=750 I have just realised that I didn't make this typo, it is actually a typo in the sample iax.conf file which is provided with Asterisk. People might want to take a look at their own settings and check if you have the same problem! Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ 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 __ http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price ___ 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] Digium connectivity issue?
They moved on Friday: Hello. We will be moving Friday, February 13th to a larger facility. Therefore, we will be closed this Friday and unable to receive faxes, ship product, or recieve phone calls. Thanks in advance for your understanding. Here is our new Address: The Atrium Building Ste. 100 150 West Park Loop Huntsville, AL 35806 Seems to be working now though -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepakumar JV Sent: Saturday, February 14, 2004 7:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Digium connectivity issue? I am having same problem and i was never successful in connecting to digium.com or asterisk.org or asteriskpbx.org for last three days. Deepak - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk-a-users-list [EMAIL PROTECTED] Sent: Friday, February 13, 2004 01:54 PM Subject: [Asterisk-Users] Digium connectivity issue? Are others seeing hugh delays and/or lack of connectivity to Digium? 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 ___ 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] X100P / Echo / ZTMONITOR CAN2,3, etc.
If you're getting echo of your own voice, but the remote is getting a clear signal, then Asterisk echo cancellation is working properly. It is the remote provider not echo cancelling properly. I don't buy it. If that were the case then why would I not _also_ get my own voice echoed with a regular phone plugged in to the same POTS line? The X101P cards are notoriously difficult to get decent audio quality out of. I know that when I used mine I tuned it so echo was minimal but the local echo (my own voice) was very fast but (to me) acceptable. I thought it was the FXS card but when I moved and simply did not have the POTS line anymore my local echo went away, with no changes to my phone/FXS card/* server. Conclusion: it was the X101P. You might ask your remote party not to use a speakerphone and to press the receiver tightly against his/her ear, as this will dampen the sound detected by the microphone and lessen the echo. Again I am almost willing to put money on ANY echo problem involving the X101P is the X101P's fault and NOT the POTS line. I say almost because you've been around these lists a lot longer than I have and I still get schooled on a semi-regular basis. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hide outgoing CallerId on Zap interface
Are you dialing out to the public network? I thinking there is a prefix you can dial out to hide your number. Is it *67? Jacques -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mickey Binder Sent: Friday, February 13, 2004 6:14 AM To: Asterisk maillist Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? 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] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Translator 'g729tolinb'
Hello, Why it occurs in log files and how it to remove? WARNING [8192]: translate.c:219 calc_cost: Translator 'g729tolinb' does not produce sample frames. Help me please... Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voip in the EU
Does anyone know where I can find some more info on the VoIP laws in the EU? Ryan
[Asterisk-Users] HELP!!!! Having problems Starting Asterisk
I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have re installed mgp123 but it still doesn't help any Ideas? Thanks in advance Robb ___ 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] Digium connectivity issue?
Yes - not much seems to be creeping out of the list servers. Iain --On Friday, February 13, 2004 07:54:50 -0600 Rich Adamson [EMAIL PROTECTED] wrote: Are others seeing hugh delays and/or lack of connectivity to Digium? 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 ___ 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_local and variables
Hi! We need to implement the following: Call comes in, ring ZAP/1 (6 rings) For the last two rings, also ring ZAP/2 [incoming] exten = s,1,DIAL(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED],18) [test1] exten = 123,1,Dial(ZAP/1) exten = 124,1,Wait(12) exten = 124,2,Dial(ZAP/2) Why not simply use this instead: [incoming] exten = s,1,DIAL(ZAP/1,12) exten = s,2,DIAL(ZAP/1ZAP/2,6) Philipp ___ 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] Hide outgoing CallerId on Zap interface
Im in the UK and unless you dial a particular code first (141) before you dial the number the phonenumber will automatically stamp the call with your main number. I THINK that this setting just stops asterisk from sending the caller ID from the originiating extention down the line (and only if it was a digital line eg ISDN) Regards Jonathan - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist [EMAIL PROTECTED] Sent: Friday, February 13, 2004 11:13 AM Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? 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] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID or Noise ?
As a new member and with less than 2 weeks experience with Asterisk, I allow myself to ask potential dumb questions... I am using a TDM400P and a X100P card in a Celeron 2,6 GHz box with 256 MB memory running RH 8.0 run level 3. In Denmark we use DTMF style CallerID sent between 1'st and 2'nd ring. Now, I've removed/diabled all CalledID settings and enabled immediate answer and this is what I get when my system answers the call *without* the mandatory s,Wait,2: -- Starting simple switch on 'Zap/1-1' -- Sent into invalid extension 's' in context 'default' on Zap/1-1 -- Executing Playback(Zap/1-1, transfer) in new stack -- Playing 'transfer' (language 'en') The question is: How do I enable any kind of debug or whatever to see the information Asterisk don't like (could it be the CallerID data) and can I capture this information and use it ? The format of the CallerID is described in ETS 300 659-1 and ETS 300 659-2 if anyone is interested, and can be found here: http://www.secret100.nm.ru/ets300659.pdf and http://www.secret100.nm.ru/ets3006590e02.pdf (I know, weird links, Google found them for me) Thanks Soren -- It is the mark of an educated mind to be able to entertain a thought without accepting it. - Aristotle ___ 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] Jitter Buffer Configuration (typo in iax.conf)
I havent been able to get the jitter buffer to work even with correct typing. I *think* I have it working decently now... What i have done is typed iax2 set jitter 250 at the CLI. Any calls after typing that seem to work decently. at least everyone in the office has not complained. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones
On Fri, 13 Feb 2004, John Bittner wrote: Anyone setup a Rhino channel bank ?... any issues. I got it working with normal pots phones but I cant get it to work with Aastra PT390 phones. The phones get dialtone but the asterisk does see any DTMF digits dialed from the phone. I had a similar problem with an adtran TA750 with digits not breaking dialtone. It would come and go, usually working fine right after a restart. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ 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] Digium connectivity issue?
I observed a packet routing endless loop at: 16 host-63-108-128-153.apid.com (63.108.128.153) This happened with traceroute from two distinct origination points. Seems to have been resolved. Message: 3 Date: Fri, 13 Feb 2004 20:11:44 -0500 From: John Fraizer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Digium connectivity issue? Reply-To: [EMAIL PROTECTED] Rich Adamson wrote: Are others seeing hugh delays and/or lack of connectivity to Digium? 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 I saw an over 12hr outage to Digium and IAXtel and I know it wasn't OUR network so, they must have had some sort of problem. John [EMAIL PROTECTED]
Re: [Asterisk-Users] Hide outgoing CallerId on Zap interface
When I need to hide callerid ( sip phones ), I will configure this in sip.conf. You need to include restrictcid=yes for each user that needs to be hidden. -- Pertti Jonathan Stanton @ Home wrote: Im in the UK and unless you dial a particular code first (141) before you dial the number the phonenumber will automatically stamp the call with your main number. I THINK that this setting just stops asterisk from sending the caller ID from the originiating extention down the line (and only if it was a digital line eg ISDN) Regards Jonathan - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist [EMAIL PROTECTED] Sent: Friday, February 13, 2004 11:13 AM Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? 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] 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_local and variables
On Sat, 14 Feb 2004, Philipp von Klitzing wrote: Hi! We need to implement the following: Call comes in, ring ZAP/1 (6 rings) For the last two rings, also ring ZAP/2 [incoming] exten = s,1,DIAL(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED],18) [test1] exten = 123,1,Dial(ZAP/1) exten = 124,1,Wait(12) exten = 124,2,Dial(ZAP/2) Why not simply use this instead: [incoming] exten = s,1,DIAL(ZAP/1,12) exten = s,2,DIAL(ZAP/1ZAP/2,6) Philipp For SIP phones (and analog phones w/ callerid), that would show two missed calls on ZAP/1 for every incoming call. Steve ___ 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] Digium connectivity issue?
I observed a packet routing endless loop at: 16 host-63-108-128-153.apid.com (63.108.128.153) This happened with traceroute from two distinct origination points. Seems to have been resolved. Guys, if they moved, then obviously this was their connection getting migrated to a new location, doesn't seem to have gone very smoothly but speaking from experience it never does, Plan C is often where things end up and too often that is no working connection, even when you plan ahead. I am sure we all have our share of nightmare stories. Lately the mailing list seems to be getting an aweful lot of bloat with people just commenting repeatedly on the same thing. Try to keep in mind other people have to sift through your responses, so if you really don't have anything to say that has not been already contributed, hold back and wait for a time you can make a difference - everyone will appreciate you then instead of grumbling. Message: 3 Date: Fri, 13 Feb 2004 20:11:44 -0500 From: John Fraizer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Digium connectivity issue? Reply-To: [EMAIL PROTECTED] Rich Adamson wrote: Are others seeing hugh delays and/or lack of connectivity to Digium? 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 I saw an over 12hr outage to Digium and IAXtel and I know it wasn't OUR network so, they must have had some sort of problem. John [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] Adtran 750 - what do I need
I bought mine off of eBay. Each cabinet should contain a PSU board (power) and a BCU (control) board. Then you have six slots for FXO/FXS cards. You'll also need the power supply (which mounts on the side.) Some units have a cabinet containing four 12V batteries. There are also two slots on for the V.35 interface, but we don.t use that. Warren H. Prince wrote: Could anyone tell me what I need to include in the purchase of an Adtran 750 to work with a T100P? Obviously, I'd need a combination or FXO and FXS boards to fit my application, but, are there any other boards that are required? Does every Adtran include the proper port to connect to the T100P, or is that needed as well? Assuming it is not included, are the FXO, FXS and T1 cards all that is needed? Does anyone suggest a particular vendor for this piece of equipment? ___ 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] Digium connectivity issue?
They moved to a different location yesterday. - Brent -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Michaelson Sent: Saturday, February 14, 2004 10:07 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Digium connectivity issue? I observed a packet routing endless loop at: 16 host-63-108-128-153.apid.com (63.108.128.153) This happened with traceroute from two distinct origination points. Seems to have been resolved. Message: 3Date: Fri, 13 Feb 2004 20:11:44 -0500From: John Fraizer [EMAIL PROTECTED]To: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Digium connectivity issue?Reply-To: [EMAIL PROTECTED]Rich Adamson wrote: Are others seeing hugh delays and/or lack of connectivity to Digium? 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 I saw an over 12hr outage to Digium and IAXtel and I know it wasn't OUR network so, they must have had some sort of problem.John[EMAIL PROTECTED]
Re: [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.
On Saturday 14 February 2004 07:39, Andrew Kohlsmith wrote: If you're getting echo of your own voice, but the remote is getting a clear signal, then Asterisk echo cancellation is working properly. It is the remote provider not echo cancelling properly. I don't buy it. If that were the case then why would I not _also_ get my own voice echoed with a regular phone plugged in to the same POTS line? Because the loop is a lot tighter in that case. There is still an echo, but because it's so quickly returned, it's not very noticeable. In the case of going through Asterisk, there is a very slight delay which makes the remote echo noticeable. I'm not sure if the Eastern European countries have upgraded in the past 10 years, but it used to be that if you called from the US to say, Yugoslavia (the specific case in which I noticed this), the party in the US would get a very bad echo. The party in Yugoslavia would hear no echo at all. In local calls within Yugoslavia, because the distance was so short, the echo was not noticeable. But in international calls, the echo was nearly unbearable (especially given the international long distance rates!). Since you couldn't hear the echo in that country, they never knew there was anything wrong. -Tilghman ___ 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] Voicemail Password Digit Timeout
On Thu, Feb 12, 2004 at 04:30:19PM -0600, Ryan R. Fligg wrote: I was wondering if there was any way to change the digit timeout or some setting of that sort on the voicemail password entry. Currently when our users enter their passwords they have to enter them very rapidly, otherwise asterisk will log the number twice. So if someone entered a voicemail password of 1234 slowly and deliberately on our system the asterisk receives it as the following number, 11223344 and thus returns the passcode invalid message. System: Asterisk CVS-02/10/04-13:27:57 built by [EMAIL PROTECTED] on a i686 running Linux 3 X100P cards 5 Snom200 phones I can't help you, but I can me too. I have a TDM400, and accessing voice-mail from these extensions is always fine. I also have a Grandstream SIP phone, and it behaves exactly as you describe. It has to do with how long the number buttons are pressed. To make it work, you have to key your PIN like the buttons are too hot to touch... I'm running the latest (.46) Grandstream firmware. I'm using dtmfmode=rfc2833 in sip.conf, and the matched setting on the phone. Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. Blessed are the censors; they shall inhibit the earth. ___ 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] Kansas SIP or IAX Provider?
Does anyone know a SIP or IAX provider for Kansasarea codes 620 and 221? Thanks! Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105
Re: [Asterisk-Users] Digium connectivity issue?
I am having same problem and i was never successful in connecting to digium.com or asterisk.org or asteriskpbx.org for last three days. We've been moving our office but we are now in the new location and hopefully won't have any more trouble. Thanks! 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
[Asterisk-Users] Is there a MaxQueueTime for Queues ?
Hi, Is it possible to kick a caller out of a queue after 5 minutes and goto the next priority in the context where they were assigned to the queue ? My desired result is that even though one agent is dynamically logged into the queue and is on a call, I would like the 2nd caller to stay in the queue for 5 minutes and then timeout to the next priority if the agent is still busy and can't get to the call. Some observations: I have tried the n option with queue (if I don't the 2nd caller will stay in the queue infefinately) eg: exten = 401,1,Queue(support1|n) The problem with using n is that with one agent logged into the queue and he is busy on a call, when the 2nd call is placed in the queue it immediately timesout and goes to the next priority in the context even if timeout=300 is set in queue.conf. Any help appreciated. -bh Here are the configs: extensions.conf [supportq] exten = 401,1, Queue(support1|t) agents.conf [agents] autologoff=15 ackcall=no ;wrapuptime=5000 musiconhold = default queues.conf general] [support1] music = default strategy = leastrecent ;context = leavemessage timeout = 300 retry = 2 maxlen = 0 This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ 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] System freeze
On Monday 09 February 2004 11:45 am, Michael Welter wrote: I have a Gigabyte K7 motherboard with an Athlon 2400+ processor. Before the T1 install I had two T100P cards, one for the channel bank and the other unused. This ran perfect for a month. Last week we installed a new integrated T1 into the unused T100P (to replace POTS lines and DSL.) In BIOS, I disabled some unused peripherals so that each T100P would find its own unique IRQ. I also installed the updated asterisk, libpri, and zaptel sources. I have seen two system freezes--one on Friday and one this morning. The whole system freezes--no LAN, no phones, no console. During this morning's freeze there were no calls in progress. The logs say nothing. Has anyone else seen this? I suspect it isn't an asterisk problem, but I would appreciate feedback. Thanks, Mike I'm running RH9 and it locks up on every kernel that I've used since before 9.0 came out. For me it's been the thing you have to live with if you want to use *. I've used completely different h/w except for the same Digium card (TDM400P w one port.) Many versions of * too. -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! ___ 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] Adtran 750 - what do I need
How you do this depends on what you're getting from your phone company. I just configured mine for incoming POTS lines. I used three quad FXO cards and sent the output out to the T-100P. Worked great, and I didn't have to change any settings on the fxo cards to get them to work. I started a thread on this very subject last month, so search the archives. Also, Bill Black was kind enough to give me the following (Thanks, Bill!) . Hi John: A few thoughts on the Adtran: If it wasn't purchased new, you may want to update the firmware if it is very old. Pretty simple to do with a kermit session or your favorite terminal program. My settings are as follows: in zaptel.conf, make sure you have: span=1,0,0,esf,B8zs fxsks=1-12 in zapata.conf, make sure you have: signalling=fxs_ks channel = 1-12 With the above settings caller ID is coming through AOK here. If your T1 lights are happy (e.g. green) and your power supply is green there is a good chance you are working OK. (ztcfg -vv should also show problems if there are any.) If you do plug into the AD-750 serial port you can also look for module status. It will hopefull show channels as being 'in service'. Watch out for a red ringer supply light on your PSU. If you see this and can't reset it just send it back to adtran. They have a 10 year warranty. Bill Black You'll probably want a serial cable for connecting to the Adtran. You'll need it for upgrading the firmware. The firmware can be found at www.adtran.com Make sure and get the L36 version of the software. Here's the how-to on connecting to the Adtran: http://www.adtran.com/ADTRANPX/Doc/0/IPT0R456V3B139RT038BE81ID8/Connecting+a+termianl+or+PC+to+the+Craft+Port.htm As you can see, there are two ways to do it. The easy way is the DB-9 male to DB-9 female straight through cable. I had a heck of a problem connecting to this until I set the flow control to off on my terminal settings. Then it was easy. You can use minicom with the VT102 settings to connect to the Adtran. Don't forget to make a proper T-1 Crossover cable to connect to the T100P! There really isn't a whole lot to getting this up and running. It was one of the easiest hardware installations I've done. By the way, I've got three quad FXS cards I'm not using, if you're interested. John Baker - Original Message - From: Michael Welter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, February 14, 2004 9:33 AM Subject: Re: [Asterisk-Users] Adtran 750 - what do I need I bought mine off of eBay. Each cabinet should contain a PSU board (power) and a BCU (control) board. Then you have six slots for FXO/FXS cards. You'll also need the power supply (which mounts on the side.) Some units have a cabinet containing four 12V batteries. There are also two slots on for the V.35 interface, but we don.t use that. Warren H. Prince wrote: Could anyone tell me what I need to include in the purchase of an Adtran 750 to work with a T100P? Obviously, I'd need a combination or FXO and FXS boards to fit my application, but, are there any other boards that are required? Does every Adtran include the proper port to connect to the T100P, or is that needed as well? Assuming it is not included, are the FXO, FXS and T1 cards all that is needed? Does anyone suggest a particular vendor for this piece of equipment? ___ 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] GSM codec with Cisco equipment
On Fri, 2004-02-13 at 22:33, John Todd wrote: I had been under the impression (after actual tests) that the GSM codec supplied with Asterisk was not compatible with the GSM codec supported by the Cisco VoIP equipment (PRI and DS3 gateways.) However, I hear recent news that this is no longer the case. Can someone with a big Cisco media gateway (58xx or similar) please test and let the community know if this works with recent IOS images? I will attempt to test with a small media gateway in the near future, but my schedule is pretty much booked solid for the next week. Using a Cisco IAD 2431 with GSM showed no problem if I use SIP, it seems that there is no support for MGCP+GSM for this box yet. -- Juanjo sin .sig ___ 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] TDM card loses Dial tone
Same here. I, too have received replacement cards from Digium, and I have even tried replacing the proSLICs, all to no avail. Also to note: the same port on each (of three) cards always goes out first. On Thursday, 12 February, 2004 19:22, John Vozza wrote: Same here... Usually after several of these show up in my system log: Power alarm on module 1, resetting! Need to unload/reload module wcfxs in order to get the dial tone back. Happens several times a week, sometimes more frequently. John - NetRom Internet Services 973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - On Thu, 12 Feb 2004, Youness El Andaloussi wrote: I experienced similar problems too with a 4 chan tdm400. This seems to especially happen when you make configuration changes. It has nothing to do with runing X or no, it does not even have to do with redhat... I experienced the same problem on mandrake. One thing you have to be extra careful is when restarting, make sure that all the modules have entirely reloaded before expecting a dialtone with an asterisk debug console asterisk -r... many of the times I thought there was no dialtone and the asterisk process had gone cukoo, I noticed that configuration was not entirely reload. Yet, reloading many times seems to get some of the TDM400 channels hung. On the other hand, this problem does not seem to happen as extensively when no reloads are made ___ 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] VideoPhone
See the video phone section here: http://www.voip-info.org/wiki-VOIP+Phones Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Isamar Maia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, February 14, 2004 1:31 AM Subject: [Asterisk-Users] VideoPhone Hi folks, Anybody knows a Grandstream-linux VideoPhone... I mean, proportionaly the same price and quality. Anybody knows? Thanks, Isamar ___ 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] Fax
Hi All, My asterisk system is running well but I can't send or receive faxes. I have an analogue fax plugged into a TDM400 connected to my ISDN 2e via an Eicon Diva. I am using G711.U - do I stand a chance of faxing or should I be doing it differently? Simon -- Simon Faulkner - Dedicated Programmes 01538 303 900 - 07771 845 326 http://dpnet.co.uk ___ 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: Rhino channel bank and aastra PT390 phones
Hey Dave, I tried that.. no change. Keep in mind that a regular pots phone works ok. Only having this issue with aastra PT390 phones. Is there something I am missing. Is the signaling different with ADSI phones. I have ADSI on in the zapata.conf I plug the PT390s into a normal pots line and they work. Anyone ever get these phone working with asterisk. Any help would be appreciated. Thanks John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis Sent: Saturday, February 14, 2004 9:33 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones On Fri, 13 Feb 2004, John Bittner wrote: Anyone setup a Rhino channel bank ?... any issues. I got it working with normal pots phones but I cant get it to work with Aastra PT390 phones. The phones get dialtone but the asterisk does see any DTMF digits dialed from the phone. I had a similar problem with an adtran TA750 with digits not breaking dialtone. It would come and go, usually working fine right after a restart. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and dial by email?
I have a friend in australia who I have set an extension up in my asterisk server that looks like so: exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]) This works fine. Now, what I would like to do is make my asterisk server ACCEPT and route calls when a user dials say [EMAIL PROTECTED] I already have _sip._udp SRV records set up for the domain I want to accept calls for. I basically want people to be able to call me just via my email address. How would one do this in asterisk? Thanks, John ___ 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] Festival: read text from external fil
Hello! I wan't to use Festival for reading text from an external textfile - anyone that has a solution for doing that? I can't figure out how I should be able to do that - if it is possible? The textfile contains the temperature and will change every tenth minute - and therefore I can't use include in extensions.conf. Best regards, Lars --- Lars Fredriksson Ockelbo, Sweden mailto:[EMAIL PROTECTED] http://www.fredriksson.net/ ___ 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: Rhino channel bank and aastra PT390 phones
I plug the PT390s into a normal pots line and they work. Anyone ever get these phone working with asterisk. Any help would be appreciated. I have a PT350 and a PT450 that work just fine in the TDM400P, ADSI and all. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax
Hi, make sure you have echo cancelation disabled on that zaptel channel. regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Hi All, My asterisk system is running well but I can't send or receive faxes. I have an analogue fax plugged into a TDM400 connected to my ISDN 2e via an Eicon Diva. I am using G711.U - do I stand a chance of faxing or should I be doing it differently? Simon -- Simon Faulkner - Dedicated Programmes 01538 303 900 - 07771 845 326 http://dpnet.co.uk ___ 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] Incoming SIP-calls and Festival
Hi! I have problems with calls that are coming from a SIP-provider, and where I want to use Festival to play som text to the caller. I hear the text if I call from a SIP-extension (I've tried with g.711a/u and GSM and all three works) But if I call in to the server through my SIP-provider I wont hear any Festival-speech (no error output on the console - see in the end of the mail), if I instead use Background for example I can hear the soundfile. I think it's very strange - is there anyone that have an idea why I can't use Festival with the calls coming from my SIP-provider. This is how it looks on the console - but the caller don't hear anything; --SNIP-- -- Executing Answer(SIP/11292-594f, ) in new stack -- Executing Festival(SIP/11292-594f, 'Hello') in new stack == Parsing '/etc/asterisk/festival.conf': Found == Spawn extension (digisip, 301, 2) exited non-zero on 'SIP/11292-594f' --SNAP-- Regards, Lars --- Lars Fredriksson Ockelbo, Sweden mailto:[EMAIL PROTECTED] http://www.fredriksson.net/ ___ 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] running asterisk as non-root
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello everyone Due to security reasons I want to run asterisk as a non root. I normaly installed asterisk, created an * user, moved the binaries to /usr/bin and chowned all the files and directories mentiont in the * manual (handbook-draft.pdf) Now I can start * but I get the following warning (which I don't get if I run it as a root): Feb 14 19:10:53 WARNING[213006]: pbx_wilcalu.c:69 autodial: Autodial: Unable to open file ~ == Parsing '/etc/asterisk/enum.conf': Found I don't know if * really works - I have't tired jet - can anybody tell me which file * want's to access? ( I looked in the source but I'm not that familiar with the code) Thank's in advance Birk Bremer -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFALoxP7QhrwFQeHVsRAq/ZAJ0VE5pGY98Ip+FlbvPYv4bHOEoXXACgkYSK m8hpZA/orrMBMRb4NoKLoJk= =7BH7 -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] Re: Voicemail Password Digit Timeout
FromJim Burwell, Dec 21,2003 __ I had the same problem with Grandsteam phones and *. No other hard or soft phones have the 'double digit' problem with *. I don't think Asterisk can do both RFC2833 and in-band DTMF at the same time. It does, however, do RFC2833 and SIP Info at the same time (SIP Info method seems to be on all the time, even when RFC2833 is selected in the sip.conf file). Switching the Grandsteam to SIP Info allowed it to talk to Asterisk and fixed the double digits problem. - Jim __ Date: Sat, 14 Feb 2004 10:56:39 -0600 From: Rob Fugina [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicemail Password Digit Timeout Reply-To: [EMAIL PROTECTED] On Thu, Feb 12, 2004 at 04:30:19PM -0600, Ryan R. Fligg wrote: I was wondering if there was any way to change the digit timeout or some setting of that sort on the voicemail password entry. Currently when our users enter their passwords they have to enter them very rapidly, otherwise asterisk will log the number twice. So if someone entered a voicemail password of 1234 slowly and deliberately on our system the asterisk receives it as the following number, 11223344 and thus returns the passcode invalid message. System: Asterisk CVS-02/10/04-13:27:57 built by [EMAIL PROTECTED] on a i686 running Linux 3 X100P cards 5 Snom200 phones I can't help you, but I can me too. I have a TDM400, and accessing voice-mail from these extensions is always fine. I also have a Grandstream SIP phone, and it behaves exactly as you describe. It has to do with how long the number buttons are pressed. To make it work, you have to key your PIN like the buttons are too hot to touch... I'm running the latest (.46) Grandstream firmware. I'm using dtmfmode=rfc2833 in sip.conf, and the matched setting on the phone. Rob ___ 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] Festival: read text from external fil
You can probably use the festival text2wave utility in a cron job to create a speech file from your source text and then use asterisk's Playback function to play it as required. Iain --On Saturday, February 14, 2004 9:41 pm +0100 Lars Fredriksson [EMAIL PROTECTED] wrote: Hello! I wan't to use Festival for reading text from an external textfile - anyone that has a solution for doing that? I can't figure out how I should be able to do that - if it is possible? The textfile contains the temperature and will change every tenth minute - and therefore I can't use include in extensions.conf. Best regards, Lars --- Lars Fredriksson Ockelbo, Sweden mailto:[EMAIL PROTECTED] http://www.fredriksson.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - oh323 - Cisco CallManager
Hi everyone, Does anyone know the answer for this situation? I have Asterisk with E1 PRI links, with SIP phones registered to Asterisk and with h.323 connection to Cisco CallManager. I am using oh323. I think I have a problem with codecs but I do not know exactly what is wrong. this is working ok: -- Call from CallManager (7960) to SIP phone on Asterisk (X-Lite or 7960with SIP image) - working OK Call from CallManager (7960) to E1 PRI trunk to PSTN through Asterisk - working OK Call from E1 PRI trunk from PSTN through Asterisk to CallManager (7960) - working OK here is the problem --- Call from SIP phone to CallManager - rings the phone, in the moment when called party picks the receiver Asterisk crashes with core dump Interesting is that if you establish a call in opposite direction (from CallManager to SIP phone) prior to that one, Asterisk wouldn't crash sometimes I will appreciate if anyone can help Tomica
[Asterisk-Users] with soekris?
does anyone has been using asterisk with a soekris 3801, if yes what distribution did you use for the soekris? thanks Kemal ___ 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] Siemens Gigaset not ringing the cordless phjone for very long
I have a gigaset 2420 connected to a DG104S. The base station rings just fine, but the cordless phones (2.4 Ghz) ring for a bout 1/2 sec and then stop. There is not enough time to pick up the line. Since this works with POTS line I assume it is how the DG104S is ringing, or is how asterisk is ringing? Or is there a setting on the Siemens that will help? -- Zot O'Connor [EMAIL PROTECTED] White Knight Hackers, Inc. ___ 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] Telian small FXS/FXO adapters
Like so many here I too still seek a small, affordable FXO gate way for 2-4 lines. I just stumbled upon the Teliann 210 online at www.teliann.com. Anyone here every use these? They seem very new as their online shop web site lists them as launched Dec 2003. They're h.323 and not SIP, but I suppose I could get around that as they're only $220 for the two port model. Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] If the gods had meant us to vote they'd have given us candidates! - Jim Hightower, Texan Populist ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE405P and dual Athlon systems
Hi, I just saw at digium.com that TE405P is out ! But it is not recomended for dual athlon systems. Anyone knows exactly why ? -- Juanjo sin .sig ___ 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 / Echo / ZTMONITOR CAN2,3, etc.
I've replaced the cable, no such luck. Thanks, Jeff From: Deepakumar JV [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc. Date: Fri, 13 Feb 2004 07:39:33 - I had similar problems and finally it turned out to be a telephone cable problem that connected from the wallmount to X100P card. But with the same cable i did not have any problem connecting a handset directly (no echo, it was crystal clear). May be try replacing the cable with a good one. Deepak - Original Message - From: Jeff Stohl [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 13, 2004 01:48 AM Subject: [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc. I have a basic x100p setup and several soft and hard phones that work great until they hit the PSTN. Like a lot of the posts I've seen I've gone through just about every echo can including mark 2, 3, and the steves with the aggressive protection on mark 2. I am running the latest CVS source code that I downloaded late last night, 0.7.2. My echo is really bad on the dialing out line only, the other user has a perfect reception. The inside user has just a horrible echo back of their voice. I've tried various echo canel methods in zapata.conf as well as regular TX/RX gains. I tried running ztmonitor 1 -v and the ztmonitor RX and TX charts came up but did not provide any display back so I was unable to find some better TX/RX. I tried several settings but those didn't help. I am configuring asterisk as a test system and was very happy with the setup until I hit the PSTN. I am hoping the T-1 setup will not have the same issues. Have I missed any steps in reducing my echo? Could my X100P card be prone to echo? Could my line really be this bad (it's not on a regular analog phone)? Can anyone recommend better echo canceling phones? I know this is a frequent topic but I truely have read through all of the past posts from the December echo can upgrade till today. Thanks, Jeff _ Check out the great features of the new MSN 9 Dial-up, with the MSN Dial-up Accelerator. http://click.atdmt.com/AVE/go/onm00200361ave/direct/01/ ___ 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 _ Get some great ideas here for your sweetheart on Valentine's Day - and beyond. http://special.msn.com/network/celebrateromance.armx ___ 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 / Echo / ZTMONITOR CAN2,3, etc.
I can hear the echo of my own voice as soon as the X100P clicks over to the POTS line to dial the call so I am positive it is not the end party. There is no echo between IP calls across country. As suggested by another user, I've replaced the cabling and tried two seperate POTS lines wired into seperate jacks. I spoke with several people in the digium IRC room who had the same problem and the long and the short of it seemed to be not to use the X100P and either suck it up and do POTS into a channel bank or go straight PRI into a T-1. I just purchased a VoiceTronix 4 port FXO card, I'll report my results on that front once complete. Thanks, Jeff From: Tilghman Lesher [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc. Date: Fri, 13 Feb 2004 20:59:15 -0600 On Thursday 12 February 2004 19:48, Jeff Stohl wrote: My echo is really bad on the dialing out line only, the other user has a perfect reception. The inside user has just a horrible echo back of their voice. I've tried various echo canel methods in zapata.conf as well as regular TX/RX gains. I tried running ztmonitor 1 -v and the ztmonitor RX and TX charts came up but did not provide any display back so I was unable to find some better TX/RX. I tried several settings but those didn't help. If you're getting echo of your own voice, but the remote is getting a clear signal, then Asterisk echo cancellation is working properly. It is the remote provider not echo cancelling properly. Think of the echo in this way: what you're hearing is your voice on the remote receiver being picked up by the remote microphone and sent back to you. You might ask your remote party not to use a speakerphone and to press the receiver tightly against his/her ear, as this will dampen the sound detected by the microphone and lessen the echo. -Tilghman ___ 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 _ Choose now from 4 levels of MSN Hotmail Extra Storage - no more account overload! http://click.atdmt.com/AVE/go/onm00200362ave/direct/01/ ___ 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] GSM codec with Cisco equipment
On Sat, 2004-02-14 at 16:47, Juan J. Sierralta P. wrote: On Fri, 2004-02-13 at 22:33, John Todd wrote: I had been under the impression (after actual tests) that the GSM codec supplied with Asterisk was not compatible with the GSM codec supported by the Cisco VoIP equipment (PRI and DS3 gateways.) However, I hear recent news that this is no longer the case. Can someone with a big Cisco media gateway (58xx or similar) please test and let the community know if this works with recent IOS images? I will attempt to test with a small media gateway in the near future, but my schedule is pretty much booked solid for the next week. Using a Cisco IAD 2431 with GSM showed no problem if I use SIP, it seems that there is no support for MGCP+GSM for this box yet. I must add. The problem of no support for MGCP+GSM is from Cisco. I have a Cisco 5300 with 12.2(something) which is controlled by BTS (no PRI but SS7) but I was playing with the config and it seems that the 5300 does support GSM but only with SIP. -- Juanjo sin .sig ___ 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] TE405P and dual Athlon systems
I'd like to know why too. I'm using a TE410P card in a dual Athlon XP system right now and it seems to be playing nicely with the dual Athlons, should I worry about something going wrong with my TE410P since it is basically the same card as the TE405P except it runs at a different voltage? MATT--- -Original Message- From: Juan J. Sierralta P. [mailto:[EMAIL PROTECTED] Sent: Saturday, February 14, 2004 8:13 PM To: Asterisk Users Subject: [Asterisk-Users] TE405P and dual Athlon systems Hi, I just saw at digium.com that TE405P is out ! But it is not recomended for dual athlon systems. Anyone knows exactly why ? -- Juanjo sin .sig ___ 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] festival voices
Thanks that works. :) So you use cepstral voices in festival? I thought cepstral was a whole seperate system. I still think the slt_arctic_hts voice from http://festvox.org/voicedemos.html sounds better then the regular cepstral voices. Brian West wrote: (Parameter.set 'Audio_Method 'linux16audio) ;(Parameter.set 'Audio_Method 'esdaudio) ;(Parameter.set 'Audio_Method 'mplayeraudio) ;(Parameter.set 'Audio_Method 'sunaudio) ; American female I'm using the cepstral frank with festival ;) (set! voice_default 'voice_frank) in /root/.festivalrc ___ 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] TE405P and dual Athlon systems
We have just one dual athlon system and we had trouble with the TE405P in the 32-bit slots of that system. We have not heard of anyone else having any problems like we had on that one system, but still had to say it i guess. The failure is that it constantly is taking errors, not the kind of thing you wouldn't notice. Mark On Sat, 14 Feb 2004, mattf wrote: I'd like to know why too. I'm using a TE410P card in a dual Athlon XP system right now and it seems to be playing nicely with the dual Athlons, should I worry about something going wrong with my TE410P since it is basically the same card as the TE405P except it runs at a different voltage? MATT--- -Original Message- From: Juan J. Sierralta P. [mailto:[EMAIL PROTECTED] Sent: Saturday, February 14, 2004 8:13 PM To: Asterisk Users Subject: [Asterisk-Users] TE405P and dual Athlon systems Hi, I just saw at digium.com that TE405P is out ! But it is not recomended for dual athlon systems. Anyone knows exactly why ? -- Juanjo sin .sig ___ 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] Xlite, GSM, Agent Logins and DTMF
Hello All, I have been testing Xlite for use with remote support agents and have run across a problem. I have Xlite setup to use gsm and rfc8322 (force DTMF inband set to no in the Xlite config). I have managed to get Xlite working with voicemail authentication but it won't accept the agents password from agentlogin. I am also noticing issues with IVR menues. In my three option main menu it will accept option 2 going to the directory, but not options 1 or 3. Xlite seems to work correctly with aLaw/uLaw. I also did some testing of the call parking extension and it would work off and on (mostly not) to transfer to extension 2000. Using Asterisk 0.7.2 -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] TE405P and dual Athlon systems
I'm using a TE410P card in a dual Athlon XP system right now and it seems to be playing nicely with the dual Athlons, should I worry about something going wrong with my TE410P since it is basically the same card as the TE405P except it runs at a different voltage? That's what I thought but apparently the drivers are different... Throwing an LDO on the board and routing 3.3V power to it would not require different drivers, so I'm not sure what precisely the differences between the TE410P and the TE405P are. Perhaps someone can explain it in further detail? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Get new PRI working
Hi all, I received my shiny new TE405P on Friday, and after much fiddling and assistance from the irc channel, I got a OK status (telco reversed the TX/RX and I wired it wrong). Anyway, currently it works for inbound calls, but I can't seem to dialout on it. Here is the config from zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17-31 dchan=16 and zapata.conf switchtype = euroisdn callgroup = 1 group = 2 busydetect = no immediate = yes context = remote signalling = pri_cpe ;stripmsd = 1 callprogress = no channel = 1-10 and here is the debug from asterisk: -- Executing Dial([EMAIL PROTECTED]:4569]/3, Zap/2/93454395||rT) in new stack Feb 15 15:58:27 DEBUG[20497]: app_dial.c:400 dial_exec: SIMPLE DIAL (NO URL) -- Making new call for cr 32774 Protocol Discriminator: Q.931 (8) len=43 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) 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: 2 ] Display (len= 7) [ Display (len= 7) [ 1 Display (len= 7) [ 1H Display (len= 7) [ 1Ho Display (len= 7) [ 1Hom Display (len= 7) [ 1Home Display (len= 7) [ 1Home Display (len= 7) [ 1Home 2 Display (len= 7) [ 1Home 2 ] Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '651' ] Called Number (len=11) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '93454395' ] Sending Complete (len= 0) -- Called 2/93454395 Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32774/0x8006) (Terminator) Message type: STATUS (125) Cause (len= 3) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (6) ] Cause data 0: 01 (1) Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) -- Processing IE 8 (Cause) -- Processing IE 20 (Call State) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32774/0x8006) (Terminator) Message type: CALL PROCEEDING (2) 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: 2 ] -- Processing IE 24 (Channel Identification) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32774/0x8006) (Terminator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Cause: Normal, unspecified (31), class = Normal Event (1) ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (Cause) -- Processing IE 30 (Progress Indicator) -- Channel 2, span 1 got hangup Feb 15 15:58:27 DEBUG[20497]: chan_zap.c:2185 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/2-1 Feb 15 15:58:27 DEBUG[20497]: chan_zap.c:1715 zt_hangup: Hangup: channel: 2 index = 0, normal = 17, callwait = -1, thirdcall = -1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: RELEASE (77) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] Feb 15 15:58:27 DEBUG[20497]: chan_zap.c:1133 zt_disable_ec: disabled echo cancellation on channel 2 Feb 15 15:58:27 DEBUG[20497]: chan_zap.c:2095 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/2-1 Feb 15 15:58:27 DEBUG[20497]: chan_zap.c:1076 update_conf: Updated conferencing on 2, with 0 conference usersFeb 15 15:58:27 DEBUG[20497]: chan_zap.c:2179 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/2-1 Feb 15 15:58:27 DEBUG[20497]: chan_zap.c:1133 zt_disable_ec: disabled echo cancellation on channel 2 -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Playback([EMAIL PROTECTED]:4569]/3, tt-allbusy) in new stack
Re: [Asterisk-Users] Kansas SIP or IAX Provider?
NANPA.NET says there is no arecode 221: Is this code reserved for future use: Yes Is this code assigned: No Is this code in use:N --- Arnold Cavazos, Jr. abcjr at abcjr . net On Sat, Feb 14, 2004 at 09:10:56AM -0800, Paul Mahler wrote: Does anyone know a SIP or IAX provider for Kansas-area codes 620 and 221? Thanks! Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 ___ 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] Digium connectivity issue?
But CVS was alive the whole time! ;) bkw On Sat, 14 Feb 2004, Mark Spencer wrote: I am having same problem and i was never successful in connecting to digium.com or asterisk.org or asteriskpbx.org for last three days. We've been moving our office but we are now in the new location and hopefully won't have any more trouble. Thanks! 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 ___ 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] Music on Hold - Context
I have set up a * box supporting 3 different companies but have some questions regarding MOH. Can MOH support multiple context or classes. Reason I ask each company would like to have different MOH sound files. Is this possible? Thanks, -gcc ___ 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] Kansas SIP or IAX Provider?
Paul forwarded the request for me. We are looking for 620 numbers for local dialing in Winfield Kansas. Most numbers are 620-221-. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Arnold Cavazos Jr. [EMAIL PROTECTED]: NANPA.NET says there is no arecode 221: Is this code reserved for future use: Yes Is this code assigned:No Is this code in use: N --- Arnold Cavazos, Jr. abcjr at abcjr . net On Sat, Feb 14, 2004 at 09:10:56AM -0800, Paul Mahler wrote: Does anyone know a SIP or IAX provider for Kansas-area codes 620 and 221? Thanks! Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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
