Re: [asterisk-users] PRI T1 Problems
Unless you're in China, you should advertise you telephone number as: +1 863 248 1195 Is your provider Quest or Qwest? Have you removed the disconnected numbers from your dial plan? [EMAIL PROTECTED] wrote: Sorry for disturbing you, but we have some problems with an installation with multiple (84) T1s from Quest. Now, our Problem is disconnected numbers are reported by sending in- band channel alert message and the B-Channel will have the tri tone and respective message but the line is never picked up and stays in ringing when dialling. So disconnected numbers are never detected as disconnected numbers. We tried this with digium and sangoma cards and the results are the same. Quest offered us a audible treatment of the line which made things even worse and would block the B-Channel for playing the message without asterisk being aware that the channel is still used. Otherwise we have a huge problem with congestion due to unavailable B- Channels when the number of connections in asterisk does not match the number of used channels / channels stay blocked after they have been hung up for several seconds, again this seems to be mostly the problems when b-channels carry audible alert messages. Thanks for any good suggestions! Frank Gorgas-Waller Explido Software USA Inc. Phone +863-248-1195Fax +863-248-1155 EMail [EMAIL PROTECTED]ICQ 7733546 --QQ- We teach penguin to fly http://www.explido.us ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE - IEEE 802.3af
You don't need to change any wiring. Just be sure that the LAN wiring terminates at a PoE LAN switch (PoE would not be passed through an intermediate switch). You will get an AC adapter with your phone. If the phone fails to power up, you can plug the adapter into the thingie in the PoE cable (not the phone). Also, the IP601 has a 24V AC adapter while the IP501 has a 12V adapter. Mike wrote: Hi, I'm not clear on how to use Power--over-Ethernet, specifically with Polycom phones. What I understand, is that by buying the Polycom 501 with the 802.3af cable bundle, I simply connect my phone, through the Polycom provided special RJ-45 cable, into a PoE capable switch, and voilà! Is this true? And if so, what happens when the Phone doesn't connect directly to the switch? (let`s say there is wiring in the wall that goes to a patch panel, for example. Do I need to change all the wiring in the office?) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Asterisk SIP Trunk and CallerID
In your SIP config for servers A and B, do you specify callerid?
Porier, Jeremy M. wrote:
on the sending server I do this:
exten = s,3,Set(CALLERID(all)=My Name1234)
exten = s,4,Noop(${CALLERIDNAME})
exten = s,5,Noop(${CALLERIDNUM})
exten = s,6,Dial(SIP/to-ServerB/${MACRO_EXTEN})
for the record, it shows the correctly set callerid and name on 4 and 5. When I do a Noop(${CALLERIDNUM}) on ServerB it shows fromServerA.
- Jeremy
From: [EMAIL PROTECTED] on behalf of Eric ManxPower Wieling
Sent: Mon 2/26/2007 5:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to Asterisk SIP Trunk and CallerID
Porier, Jeremy M. wrote:
I have two asterisk boxes (1.2.14) connected via SIP with Polycom 501s
registered to each. I set callerid name and num before sending the call
from one box to another but the phone registered to the receiving server
only properly shows the caller name, not the number. The number on the
phone always shows as the name of the sip registration of the calling
server.
Do I have to set a seperate sip header in the dialplan if I want to pass
callerid name and number between two boxes? I feel like I'm making this
too complicated.
Show us the line that sets the Caller*ID in your dialplan.
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--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
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Re: AW: [asterisk-users] ReceiveText()?
What is the mechanism used to SEND TEXT over a Zap channel? Is it FSK? Roland Ndaka Fru wrote: Here is how you can send/receive text in the DialPlan using an AGI script: print STDERR 1. Testing 'sendtext'...; print SEND TEXT \hello world\\n; my $result = STDIN; checkresult($result); print STDERR 2. Receiving Text 'receivetext'...; print RECEIVE TEXT 3000\n; my $result = STDIN; checkresult($result); Greetz, Roland. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Olle E Johansson Gesendet: 24 February 2007 10:52 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] ReceiveText()? 24 feb 2007 kl. 03.15 skrev Yuan LIU: How do I receive text sent from SendText() application? Asterisk lists text capability, so SendText() is successful. But I don't see an application to actually use it. EyeBeam and several SIP phones does receive those messages. We need to make sure that the application and the parser supports UTF8 messages, as both SIP and IAX2 is standardized on UTF8 text messaging. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for starting point?
Go to your book store and get the Fedora/Linux reference. Get yourself a PC with 20GB drives, a CD burner, and decent ram. The PC should have either an i386 or x86_64 processor. If you'll be purchasing a PC, go to the computer store, purchase the piece parts, and assemble it yourself (I like Athlon CPUs). Since you have a PC and were able to post a message, go to http://fedora.redhat.com and follow the links to Documentation and then to Download. Follow the link for your PC's architecture (i386 or x86_64) and then download the six ISO images. Burn each image to a CD. Install Linux. Take all the defaults. Load all packages. When it's running, login as root and open the browser. Go to http://asterisk.org . Take the download tab and download the five Asterisk 1.2 tar files into directory /usr/src. 'cd' to /usr/src. Use 'tar xzf file' on each of the downloaded files. Enter the zaptel directory and execute 'make', check for errors, and then 'make install' and 'make config'. Enter the libpri directory and execute 'make;make install'. Enter the asterisk directory and execute 'make', 'make install', and 'make samples'. Enter the asterisk-addons directory and execute 'make' and then 'make install' Enter the asterisk-sounds directory and execute 'make install'. Execute 'service zaptel start'--this will load the zap drivers. These will also load on reboot. Execute 'asterisk -c'. This will start Asterisk. N.B. People spend years between step #1 and a running Asterisk system. For help with asterisk, google on 'site:lists.digium.com search words'. For the wiki, google on 'site:voip-info.org search words'. The wiki is most helpful. Keep a blog of your experience and let other newbies learn from you. :=) Cheers, Gary H. Thompson wrote: Hi, I am a retired telephone tech/manager who recently had a bad experience with a local company offering digital phone service (VoIP). I have spent the last thirty years in the PSTN network, switching, PBX and key system field and am interested in learning more about VoIP. My background also includes programming, mostly specialized applications to interact with the PSTN network. Most of my experience in this field have been with Borland products, specifically Delphi. I also have been involved with database programming, same platform as the communications. My computer experience started with the operating system CPM (I’m not really that old, only 56). The best platform now seems to be ƒ so now since I am retired now, it seems a good time to learn something new. I also have been looking at Asterisk which most companies seem to be using for a PBX platform. I found out by accident that the local company I had the problem with uses this PBX software. Could someone steer me in the right direction as to where to start? I spent most of my career in the telephone industry in a ‘bush’ area of Alaska so pretty much had to teach myself what I needed to know about computers but I can learn almost anything from a book and by asking questions when I get stuck. Most of my experience was before the Internet so I plan on using this avenue to advance my knowledge. I understand what a broad scope I am asking about so would appreciate any tips to help me get started. Since there are many ‘brands’ of Linux what is the best one to start with? Which Linux will be better when I get to the point of working with Asterisk? Any tips or ideas on book s, online tutors, discussions or anything of this nature would be much appreciated. I hope to add to this group if I can be any assistance from the ‘other side’, the PSTN network. Thank You, Gary H. Thompson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Can anyone help me out with Polycom 2.1 firmware please?
I can provide Polycom phones, and I have provisioning scripts. Is that what you need? Eric Bishop wrote: Any kind Polycom dealers out there? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diagnosing poor call quality
The advertised datarate (8mb/448k) are the speeds at which the circuit between the customer and the central office is clocked and has no relationship with *effective* throughput. At the central office are *shared* facilities than connects each DSL connection with the network, and over subscription to these shared facilities cause congestion. Also, there is no QoS on the Internet, and congestion anywhere between the end points will cause poor call quality. Disclaimer: The following information is several months old--I've since moved my customers away from Qwest DSL. Here in Denver we have Qwest DSL service from a central office where the effective throughput drops to dialup speeds during the day. Regular web/email users don't usually notice packet loss because dropped packets are recovered by the TCP protocol. For VoIP on UDP, however, the call quality suffers to the point of being unusable (clicking, popping, and dropouts). Furthermore, Qwest doesn't have Denver peering with the rest of the Internet. To leave the Qwest network, connections typically go to DAL, LAX, or SFO on congested circuits. So beware of VoIP over DSL. Your users need to be aware of the tradeoffs between the cost of DSL vs. T1 and the effect on call quality. Chris, if your customers are in the western US then please contact me about dedicated circuits. Chris Bagnall wrote: Greetings list, We have an issue with call quality at 2 sites where the users (4 Elmeg IP290s at one site, 2 SPA942s at the other) do not have an asterisk box on-site. Each site has an 8mb down/448k up ADSL connection and the phones connect via SIP to an asterisk box in a datacentre using g729. The asterisk box in the datacentre connects to our other asterisk boxes providing pstn connectivity via IAX2. Latency between these boxes is between 1 and 2ms. The ADSL connections to the client sites are all consistently delivering latencies of sub-25ms to the datacentre and there is traffic shaping on that connection to give priority to any traffic from the phones' IPs. Comments from the users at these sites are as follows: call sounded like a dalek and I couldn't make out anything the caller was saying the phone on my desk is breaking up so badly it's virtually unusable calls sound like they're breaking up with metallic background noises We have quite a few customers with asterisk boxes on-site (with phones connected to them via the LAN) using ADSL connections from the same supplier, and are not having these issues with them. canreinvite=no and nat=yes are set on all these devices, since they are behind NAT. Each device re-registers with asterisk every 5 minutes to prevent any possible NAT state timeouts. Any pointers/places to look for potential problems would be much appreciated. Regards, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One-way audio after several minutes 1.4.0
Three sites are experiencing ~10sec period of one-way audio. This happens several minutes into the call, and it is very intermittent (infrequent). It does not happen on inter-office calls but only on calls to/from the PSTN. Occasionally, a spurt of white noise precedes the drop-out much like a cell phone drop-out. One site uses one port of a Sangoma A102X (PCI-X). Another site uses a TE110P. The third site uses a TDM04B (with a TDM10B for the fax machine). All sites operate normally except for this intermittent problem. Each machine has its USB drivers removed, and there is no sharing of interrupts. I've set interrupt latency on the NICs down to 32. These machines are dedicated to Asterisk, and there are no other server processes (web, mail, etc.). I was thinking interrupt starvation on the PCI cards. We upgraded one site from a Dell desktop to a Dell server (with A102X), and the problem mostly went away but did occur twice on Monday. The commonality between all sites is Asterisk/Zaptel 1.4.0. The TDM04B site started reporting this problem after the upgrade to 1.4.0. Thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco SmartSwitch
Is anyone having problems with Cisco's 2960/3560 LAN switch? Problems causing retries exceeded in Asterisk? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One-way audio after several minutes 1.4.0
More info: I've noticed that Asterisk CPU utilization has spiked to 100% for a period of 10-20 seconds. Michael Welter wrote: The commonality between all sites is Asterisk/Zaptel 1.4.0. The TDM04B site started reporting this problem after the upgrade to 1.4.0. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP phones at multiple locations
Each employee has a Polycom phone at his desk at the real office as well as a Polycom at his home office. I'd like a call to the employees extension to ring both phones. I'd also like one entry in the buddy list for each employee, and the buddy list to indicate he was on a call no matter which phone was being used. Should I do this in the dial plan, or should I register each phone on the same SIP userid? Has anyone done this? Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Power Specs
The 501 is 12VDC, and the 601 is 24VDC, as I recall. There was a post a few months ago that said that plugging the 24VDC into a IP501 will fry the phone. Peder @ NetworkOblivion wrote: Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recordings.
Has anyone tried recording to a ramdisk? To an NFS mount? Was there a benefit? [EMAIL PROTECTED] wrote: Hi, We want to build an Asterisk system that needs to be able to record, when in a peak situation, a maximum of twenty calls simultaneously. I could not find any reference to performance and recording. I need to order a new server but need to know the specs I need. Does anyone have experience with recording multiple calls simultaneously on a single system with or without performance trouble? What kind of system do I need? John Vermeeren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recordings for VR analysis
Is there a programmatic to to trim the silence from the beginning and end of a recording? From a .wav file? From a .ulaw file? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDD/TTY device for the deaf
I have a client who would like to install a TTY device in their office. They have a Asterisk PBX with a CBeyond T1 to the PSTN, and the system uses the CBeyond SIPConnect facility for PSTN voice calls. The TTY device would connect with a TDM10B (FXS) card and have its own DID number. I believe the protocol between the calling TTY and the local TTY is FSK. In any case, would two TTY devices be able to communicate over an RTP stream using g711? Or should the TTY device be attached to a POTS circuit away from the PBX? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lumenvox speech recognition
Does anyone have experience with this product? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attended transfer hanging PRI channel
It happens both when the hold button is pushed and when not pushed. Blind transfers seem to work properly. Doug Lytle wrote: Michael Welter wrote: The attendant attempts an attended call transfer (all phones are IP501). The attendant pushes hold, transfer, dials the extension and announces the call. When the attendant pushes transfer the second time, the original call is lost. The attendant is doing it incorrectly. Pressing the hold button isn't necessary. Pressing transfer will automatically put the call on hold. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attended transfer hanging PRI channel
The attendant attempts an attended call transfer (all phones are IP501). The attendant pushes hold, transfer, dials the extension and announces the call. When the attendant pushes transfer the second time, the original call is lost. The reason this is a big problem is that the PRI channel for the call remains busy. Subsequent inbound calls on that channel are rejected. Asterisk 1.2.12.1, Polycom SIP 1.6.6. Has anyone seen this? Thanks. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new in 1.4?
The docs say that SMDI works only with an FXS interface. What does that mean? What if Asterisk was a voice mail system on a legacy PBX, connected via a T1 and SMDI? Joe Pukepail wrote: I seen something in the bug tracker and svn about SMDI. Not sure if it was included it 1.4 though. Would be interested if anyone knows if this will work with nortel system (option 11 in particular). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] University switches to Asterisk
Yes. I don't use my customer's names on the list, so I can't say anything. Porier, Jeremy M. wrote: They're not the only ones :-) Jeremy Porier Senior Director of Information Systems and Technology Colorado Christian University [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, September 13, 2006 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] University switches to Asterisk Interesting article I found linked from Groklaw: Sam Houston State University replaces Cisco CallManagers, Nortel PBXs with Linux-based VoIP and messaging servers http://www.networkworld.com/news/2006/091206-von-sam-houston.html?page=1 Doug -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: PRI: sometimes Asterisk drop calls
Do you have queues/agents configured? Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... To start the ball rolling: Software: Zaptel 1.0.9, Asterisk 1.0.9, BRIstuff-0.2.0, wanpipe-2.3.2 PRI interface: Sangoma A101U (UK E1) Phones on sites with NO problems: snom, elmeg, Aastra, Linksys/Sipura Phones on problem site: Hitachi WIP3000, Zyxel F1000 (?) Hi Steve! I have same errors on one *. Here are basic information's: Sep 5 15:06:53 WARNING[2223] chan_zap.c: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. Sep 7 12:56:10 WARNING[2224] chan_zap.c: Ring requested on channel 0/1 already in use on span 2. Hanging up owner. Software: asterisk-1.2.10; libpri-1.2.3; zaptel-1.2.7 PRI interface: Digium Te205P (HR T-com E1) Phones: No phones - connected to Ericsson E250 BP -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: PRI: sometimes Asterisk drop calls
One of my clients is saying that this happens after a queue agent performs an attended transfer. Has anyone else seen this? Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... To start the ball rolling: Software: Zaptel 1.0.9, Asterisk 1.0.9, BRIstuff-0.2.0, wanpipe-2.3.2 PRI interface: Sangoma A101U (UK E1) Phones on sites with NO problems: snom, elmeg, Aastra, Linksys/Sipura Phones on problem site: Hitachi WIP3000, Zyxel F1000 (?) Hi Steve! I have same errors on one *. Here are basic information's: Sep 5 15:06:53 WARNING[2223] chan_zap.c: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. Sep 7 12:56:10 WARNING[2224] chan_zap.c: Ring requested on channel 0/1 already in use on span 2. Hanging up owner. Software: asterisk-1.2.10; libpri-1.2.3; zaptel-1.2.7 PRI interface: Digium Te205P (HR T-com E1) Phones: No phones - connected to Ericsson E250 BP -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI channel hangup
There was activity in late 2005 concerning PRI channel lockups. The telco sends a call to channel n, but Asterisk thinks channel n is busy and rejects the call. There was an entry in the bug tracker and chatter on the list. Has this problem been resolved? I have two accounts experiencing periodic channel hangs--one account on an Eschelon PRI and the other on a Nortel PRI. One account is on Asterisk v1.2.9.1 and the other v1.2.7.1 . Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI channel hangup
This seems to happen when an agent makes an attended transfer. Does anyone have more information? Michael Welter wrote: There was activity in late 2005 concerning PRI channel lockups. The telco sends a call to channel n, but Asterisk thinks channel n is busy and rejects the call. There was an entry in the bug tracker and chatter on the list. Has this problem been resolved? I have two accounts experiencing periodic channel hangs--one account on an Eschelon PRI and the other on a Nortel PRI. One account is on Asterisk v1.2.9.1 and the other v1.2.7.1 . Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Deadlock
(I'm getting 404 Not Found from the search engines) I have a system that gets a deadlock every week or so. On the logs I have many channel.c:787 channel_find_locked avoided deadlock for 0x837730 messages. The system has an Eschelon T1 with 6 voice (with dchan) arriving on a TE110P. Asterisk 1.2.7.1, Linux FC5. The system also has a TDM30B card. The phones are IP501. During the deadlock period, outbound calls are ok. However, an inbound call (on channel two) is rejected because Asterisk thinks the channel is in use. There are no call queues on this system. I see a deadlock bug, but it has to do with queues. Can someone shed some light on this situation? Thanks, P.S. This system often fails to reboot properly. zaptel doesn't load correctly, and Asterisk goes into continual restart. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk server crashes after two years
My Asterisk colo server has been up for almost two years. Today it crashed. When I gave the reboot command, it crashed so hard that it had to be power cycled. I wasn't in attendance, but I can speculate that it had a kernel panic during the shutdown. Yesterday I added a PHP agi script, and it had been user over 1000 times before the crash. I don't think the Linux/Asterisk crash is coincidental. Can someone give me things to look for? I'm watching memory, and it has 750MB free (out of 1GB). When I restart Asterisk, I see 19 processes--is this normal? What else should I be doing to narrow down on this problem. Thanks for your help. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom config error 0x4020: possibly related to RE:Polycom upgrade issue?
I would get the same error when trying to use sftp. Switching to ftp eliminated the problem. Curt Shaffer wrote: I posted earlier about an application not found error. I have manually pointed the phone at the server but it just does not seem to ever even hit it. I am going to do some network captures here soon after I walk away from this computer for a while. But here is another question which I am not sure if it may be related. After loading the application successfully on other phones I get config error 0x4020 and it just keeps rebooting through this whole process. I have checked my configs and checked them twice against all documentation I could find, and from what I see they are OK. I have posted one here for you all to look at and maybe you can see something I am missing. MAC.cfg (located in /ftproot/ ?xml version=1.0 standalone=yes? !-- Default Master SIP Configuration File-- !-- Edit and rename this file to Ethernet-address.cfg for each phone.-- !-- $Revision: 1.13 $ $Date: 2004/11/26 23:30:44 $ -- APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=x102/x102.cfg, sip.cfg MISC_FILES= LOG_FILE_DIRECTORY=x102/ ## X102.cfg (located in /ftproot/x102) ?xml version=1.0 standalone=yes?^M PHONE_CONFIG OVERRIDES reg.1.server.1.expires=60 reg.1.address=102 voIpProt.SIP.outboundProxy.port= log.level.change.cfg=0 _.0x20._log.level.change.sip=0 log.render.level=0 tcpIpApp.sntp.gmtOffset=-21600 tcpIpApp.sntp.address=xxx.xxx.xxx.xxx reg.1.server.1.address=xxx.xxx.xxx.xxx reg.1.auth.password=1234 reg.1.auth.userId=102 voIpProt.server.1.register= reg.1.displayName=Test voIpProt.server.1.address=xxx.xxx.xxx.xxx reg.1.ringType=8/ /PHONE_CONFIG I also have a .cfg file in this directory that has the following: ## .cfg ?xml version=1.0 standalone=yes? !-- Default Master SIP Configuration File-- !-- Edit and rename this file to Ethernet-address.cfg for each phone.-- !-- $Revision: 1.14 $ $Date: 2005/07/27 18:43:30 $ -- APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=phone1.cfg, sip.cfg MISC_FILES= LOG_FILE_DIRECTORY= OVERRIDES_DIRECTORY= CONTACTS_DIRECTORY=/ Any help would be appreciated. And I realize this is more of a Polycom question rather than an Asterisk question so if anyone can point me to a good polycom list I would appreciate it as well. Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI record_file
From the Eyebeam softphone, is there a way to capture the AVI stream to a file? I tried PHP record_file, but it seems to want an audio file. Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom, TFTP, and DHCP
Kevin Smith wrote: Michael, Maybe I am not understanding your question, are you saying that when you configure your phone with a static IP address, you cannot find the boot server and when in DHCP you can? The phone uses DHCP to get its IP address. In the phone's server params, I enter the IP address of the tftp server. Without the next-server entry in the DHCP configs, the phone says it cannot find the boot server (and uses the previous configuration). However, when next-server in DHCP is set with the tftp IP, the phone loads its configuration from tftp and boots normally. I'd like to not have to set the tftp address in DHCP, because I don't always have access to the DHCP server. Is there someway to tell the phone to override the DHCP server setting? Is there something I'm missing with the phone's network config? Thanks If you are having problems with the phone having a static IP address, make sure it is getting the correct IP, subnet, gateway and DNS. If your DNS is incorrect for example, you won't be able to find the server you entered, since there will be nothing to point the phone where to go. If you are talking about the actual boot server location, that needs to be static as far as I know. It isn't like DHCP addressing where it gets the DNS information from the host. It's a parameter that needs to be set. If your TFTP server is changing IPs I would strongly suggest giving it a static IP. It will make your life a lot easier. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom, TFTP, and DHCP
When I set the tftp address into the IP501 server parameters and boot, the phone says it says it cannot find the boot loader and reuses the previous configuration. When I set the tftp address in DHCP and reboot the phone, it finds the tftp server and loads correctly. My problem is that I don't always have control of the DHCP server. Is there a way to set the phone to find the tftp server on its own? Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 601 question
Are you using PoE? I had two PoE IP501s on long LAN runs, and the phones would periodically reboot. I plugged an AC adapter into the phone (with PoE) and everything is fine. Kevin Smith wrote: Hey everyone, I know this isn't a direct Asterisk issue, but some of you may know this answer. I recently upgraded the SIP version to 1.6.6 on all of our phones in the office. Everything is working fine, except one aspect. The phones in the office reboot randomly for no apparent reason. I haven't changed anything in the configuration files since the upgrade. The only setting in the sip.conf file that I can think would cause this problem is voIpPort.SIP.specialEvent.checkSync.alwaysReboot=0 Which is to me is fine, I wouldn't want the phones to reboot unless I did change something in the configuration files. Any other thoughts as to what may have caused the phone to reboot? Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls
Look in the sip.conf (or whatever) and make sure the context specifies a context that allows outgoing calls. Leah Newmark wrote: Just saw this message back to me...I didn't realize my message was posted and I ended up posting again. Oops! Anyway, she *is* able to receive calls. She gets a fast busy when trying to dial anything. I know we had her do speed tests on her DSL the end of last year but I don't remember the outcome, but that was for quality issues, so I don't think it has to do with this problem per se. Any ideas of what to test or look at? Thanks! LN Message: 18 Date: Tue, 20 Jun 2006 00:12:51 +0200 From: lenz [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] User Loses Ability to Make Outgoing Calls To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; delsp=yes; charset=iso-8859-15 Hello Leah, it may be the quality of her link degrading - it happens easily with ADSL. which error does she get? and she cannot receive calls at the same time, right? l. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to use a data T-1?
Is anyone using the HDLC facility in Zaptel to bring a data T1 into an Asterisk system? I know this was available in kernel 2.4.19--is anyone using it in kernel 2.6.x? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH too loud
I ripped a rock-and-roll CD for a client's moh. But it's too loud. Is there a simple way to reduce the gain without having to remix the tracks? Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GXP-2000 (steer clear)
Is there a way to reduce the gain on the handset mic? Erick Baum wrote: So far, pretty much every firmware upgrade has been an improvement in one way or another. They're running 1.1.0.13 http://1.1.0.13 now which seems to be fairly stable. They still get brief moments of echo, but usually only when the handset volume is up pretty high. I think it's a hardware problem where the handset microphone is actually picking up the handset speaker audio when it gets real loud. If you do get these phones, I would make it very clear to your customers that this is a very inexpensive phone, aka CHEAP. So don't be surprised if you have some problems every once in a while. But they do have some great features that you only find in more expensive phones and some features even the big boys don't have. They especially work well remotely, extremely nat friendly... which I cannot say for the Cisco's or Polycom's. Erick On 6/7/06, *Louis-David Mitterrand* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Tue, Jun 06, 2006 at 11:26:20PM -0400, Daniel Salama wrote: Well, these are encouraging words :) You're basically telling me that I should tell my client to buy other phones. I agree that you cannot compare these phones with Cisco or Polycom. After all, like you said, what do you expect for under $90. However, the fact is that my client just recently invested in these and it will be hard, if not impossible, for me to tell my client to swap them for Polycoms or something else at a much higher cost. I have heard complaints from my client about the speakerphone and they are now, I guess, getting used to picking up the handset :). I have heard any echo problems so far. What bothers me the most is that the phone stops working often (multiple times per day). By this I mean that my client won't be able to dial anything successfully. As soon as 3 or 4 digits are entered, they get a fast busy. To solve it, they need to reboot it. It sounds as if these phones were running Windows instead of Linux :) Anyway, what firmware did you use that solved so many of your problems? I've only had bad experiences with these phones and steer clear of them. In the same price range you can now get the Thomson ST-2030 or Polycom 430 for a much, much better user experience. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Delayed Answer
I have an Asterisk system connected with a CLEC that provides SIP termination. When placing calls from phones on the Astersik system to the PSTN, the calling party hears ringing while the called party is saying hello. The problem appears to happen when calling a POTS line. The problem does not seem to occur when calling a PSTN number on a T1 circuit, nor does it occur on inter-office calls. When I listen to the Monitor recordings, I hear ringing and then the called party saying hello? hello? hello? while the calling party hears ringing. Is the ringing that the calling party hears generated by Asterisk and not the ringing in the received audio stream? Would this problem occur if the 200 (OK) message were delayed for any reason? Does Asterisk wait for the 200 message before it connects the received RTP stream with the calling party? Thanks, Asterisk v1.2.4 Polycom IP501 phones Private network between the Asterisk system and the CLEC. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing Calls Not Working all the time
Some central offices cannot immediately accept digits. Try preceding your dial string with a 'w'. Andrew Berman wrote: I currently have Asterisk 1.2.7.1 http://1.2.7.1 and the Sangoma A200 w/ 6 FXO ports and HW Echo canceller. I have outgoing calling setup to use a group so that if one channel is busy it goes to one of the other channels. What's weird is that when I dial an outside number, sometimes it goes through and other times I get You have reached an invalid pager number MCLL327. I have no idea what that means. Is Asterisk calling the wrong number sometimes, or is there something wrong with the phone line? Is there any sort of Asterisk config I can use to avoid this issue? Thanks for any help you can provide, Andrew zaptel.conf: loadzone = us defaultzone = us fxsks = 1-6 zapata.conf: [trunkgroups] [channels] echocancel = yes echocancelwhenbridged = yes rxgain = 0.0 txgain = 0.0 useincomingcalleridonzaptransfer = yes callreturn = yes callwaiting = yes threewaycalling = yes cancallforward = yes busydetect = yes busycount = 6 musiconhold = default callgroup=1 pickupgroup=1 signalling=fxs_ks context=incoming1 group = 1 channel = 1,3-6 context=incoming2 group = 2 channel = 2 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spam
I was just spammed from exoxshaindia.com. The spoofed from address was asterisk-users asterisk-users@lists.digium.com, and the subject was Fw: Real show and it containd attachment 3.92315089702606R02.UUE. I believe UUE is a compressed executable. WATCH OUT! -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 resource full problems ...
My customers are reporting that the contact directory can only hold about 45+ entries. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My consulting story
When I review all the proposals I have submitted, I realize that I hardly ever sell to a prospect that has an IT staff, especially a staff of one guy who knows absolutely everything. Now my proposals to such prospects are nothing more than a brief description of functionality and the pricing--no more DIY manuals. I no longer mention brand names or my supplier names. I've also begun requiring an up-front payment for additional equipment, especially telephone sets. Some customers ask for terms, to which I say Does Home Depot give you terms?. I've been able to significantly reduce my receivables with this policy. When consulting with a company with an established presence in the community, you really have to give then Net30, especially if they have a formal accounts payable policy. For telephone consulting, you first need the client to demonstrate his willingness to pay, even if its just $100. Technical Support wrote: That's the nature of consulting - you have to balance demonstrating competency with solving the problem before being paid. We've had many similar experiences, and we now require prepayment for 2 hrs service before we do any work (or even talk to the client for more than a few minutes). (Despite attempts by potential clients to make the sales call into a problem solving call). We have undoubtedly lost potential opportunities, but our walk away with free advice effort has been almost eliminated. The same goes for proposals. I can't count the number of times our proposals have become a do-it-yourselfer's guide to setting things up by themselves. I think it's great if customers want to do it themselves, but don't waste our time! I understand of course that it's tough for users too. There are lots of self proclaimed experts on the list who after hours of billed time have done nothing for their money (we've cleaned up after lots of those folks too). These are usually the same people sending out flame emails about how smart they are and how stupid everyone else is. MD *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Voce Lavoce *Sent:* Friday, April 14, 2006 5:14 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] My consulting story Hi everybody, I would like to be awareabout what happened to me. Two weeks ago, on a Sunday morning a French guy called me. Ask me to fix some problems with his asterisk. After fixing his problem, he asked more and more, after 10 hours of work I ask him to pay me for the first milestone. However, lucky me that I did not finish, since he never paid me. Be afraid and take your action if some french guy wants to hire you to do some trunking with the Philippines. Hope, that this can help someone. See you ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Jitter in SIP connection
I'm experiencing a very strange problem with SIP calls with a CLEC (CBeyond). The downstream audio with the telephone on mute is excellent. However, when there is upstream audio (i.e., breathing) from the mic, the downstream audio is clipped and sometimes dropped. The strange thing is, if I Monitor the call, the downstream audio in the wav file is perfect, even though there was clipping and drop-outs in real time. Is this a case of jitter? What are the symptoms of jitter? Does jitter resolve itself in the wav file? Does chan_sip have a jitter buffer yet? When I move the calls to another ITSP, I don't have clipping and drop-outs, so I'm assuming the problem is not with the Asterisk system or the telephones. The Asterisk version is 1.2.5. The phones are Polycom, Cisco, and Grandstream. I've checked my NIC connections and everything is full duplex. Thanks for your help. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Jitter in SIP calls?
I'm experiencing a very strange problem with SIP calls with a CLEC (CBeyond). The downstream audio with the telephone on mute is excellent. However, when there is upstream audio (even breathing) from the mic, the downstream audio is clipped and sometimes dropped. The strange thing is, if I Monitor the call, the downstream audio in the wav file is perfect, even though there was clipping and drop-outs in real time. Is this a case of jitter? What are the symptoms of jitter? Does jitter resolve itself when the call is recorded? Does chan_sip have a jitter buffer yet? When I move the calls to another ITSP, I don't have clipping and drop-outs, so I'm assuming the problem is not with the Asterisk system or the telephones. The Asterisk version is 1.2.5. The phones are Polycom, Cisco, and Grandstream. I've checked my NIC connections and everything is full duplex. Thanks for your help. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incorrect CDR results
When I look at my CDR data for calls to NuFone, the billsec for each call is 14 seconds or less. When I look at my NuFone account, the billsec has normal call lengths. So it seems that the billing on the Asterisk system terminates after about 14 seconds. The calls come in on an IAX connection and go out to NuFone on IAX. Are these calls bridging away from the Asterisk server? How can I get accurate billing data? I tried to Google the archives but I'm still getting page not found. Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: How is Teliax ?
You cannot criticize Teliax until you investigate how your calls are getting to them. I have a customer on 17th St. in downtown Denver who use Qwest.net as their ISP. They use Teliax (on 16th St.) as their ITSP. Piece of cake, right? This may have changed recently, but Qwest doesn't have any peering arrangements in Denver (!), so, to get to the rest of the Internet, Qwest traffic is routed over a very congested circuit to Dallas where it has a peering arrangement with Sprint. For the ordinary Internet user, this trip to Dallas using TCP won't be noticed. The TCP protocol will resend any dropped packets. For my VoIP customer having UDP packets dropped at congested routers in Dallas, it's a disaster. This VoIP connection between Qwest.net and Teliax is not suitable for VoIP, and it's not the fault of Teliax. Having said all that, I see where Teliax have installed the voip-co4 host on Viawest. Are you using that host for your analysis? [EMAIL PROTECTED] wrote: Lonnie Abelbeck wrote: asterisk at anime.net writes: On Thu, 30 Mar 2006, Giridhar Reddy Bandi wrote: I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on Teliax before i purchase. suggest me if there are better sevice providers. I have had issues with termination on teliax. Callers tell me I sound choppy to them. Teliax origination has no problems at all strangely enough. If you used SIP instead of IAX2 with Teliax you will have better quality calls. The 'choppy' sound occurs with IAX2 and not SIP at Teliax. I can recommend Teliax, but use SIP. But I _am_ using SIP. I tried all the various teliax gateways including the beta test ones and had choppiness with all of them. As I said before, teliax origination had no choppiness problems at all. Only termination had issues. I had no problems - termination or origination - with junction networks, despite the fact they had 3x higher latency than teliax. JN is more expensive than teliax though. Also, I have talked to others who had similar choppiness problems with teliax. So it's not just me. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI issues
Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 17 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 18: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 18 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 19: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 19 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 20: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 20 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 21: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 21 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 22: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 22 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 23: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 23 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple auto attendants
We're assuming you will use a T1 (or E1) for your PSTN interface. If you're using POTS lines then there will be no information about which number was called--you'll need a separate POTS line(s) for each tenant. We have multiple tenants on our hosted PBX without problem. I was given the task to try and build a VOIP solution to an office building with multiple tenants in all sizes and shapes. Some of them will require auto attendants and some will simply want direct lines to their phones. The question I have is: Can asterisk be configured to handle multiple auto attendants? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: How is Teliax ?
And, what does traceroute say about your connection with Teliax? Hmm? [EMAIL PROTECTED] wrote: On Fri, 31 Mar 2006, Michael Welter wrote: Having said all that, I see where Teliax have installed the voip-co4 host on Viawest. Are you using that host for your analysis? I have used every single gateway teliax has made available to me, including their beta test ones. I experienced choppiness with all of them, but only when terminating calls. As stated before, origination is perfect. With junction networks -- no problems whatsoever with either termination or origination. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP - Problem with audio clipping
Using a SIP connection with a CLEC, the downstream (received) audio is perfect when the mute button is activated on the phone. However, when there is upstream audio (i.e., talking or even breathing into the microphone), the downstream audio is cut off. It's kinda like having a half-duplex audio connection. When I divert outgoing calls to another provider, these calls are fine. However, the inbound calls from the CLEC continue to have clipped audio. This happens with Polycom, Cisco, and Grandstream phones, so I don't believe it's a phone problem. I have this problem with two different Asterisk system using the same CLEC. One is * v1.2.4 and the other is v1.2.5. The systems have totally different motherboards. Has anyone had a similar problem, and what was the cause? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not Found in archive
I'm seeing quite a few Not Found pages when I google lists.digium.com. Is anyone else getting this? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP501 Buddy List
I have a problem with my Polycom phones. In the buddy list, the phone displays all but three employees. For those three employees, there is no difference in any of the configurations. Is there a secret to getting all employees into the buddy list? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Testing IAX links
I need to test QoS on an IAX link between a server in Colorado and a server in Europe. I know I could install a Milliwatt extension on the European server and just listen, but is there a more scientific method to collect QoS metrics? Thanks P.S. I'm getting a lot of Page Not Found on lists.digium.com. Are the older posts being purged? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog Desktop Phone
My customer with Cortelco phones is *very* unhappy. They expected Polycom speaker phone quality on a crap $50 phone. The only reason I didn't have to take the phones back is because the IT guy that ordered this system quit the company. My first Asterisk system dates back to early 2003. That customer has an Adtran 750 w/ Office Depot analog phones. That system is still in place, unchanged. Today, I could install that same system at a lesser price using decent SIP phones. With the GXS2000 at $82, what justification is there to install analog phones on a channel bank? If you have unlimited time to deal with an aggravated customers, then go for the Cortelco/analog solution. If you want a happy customer, install SIP phones. Today, I installed a dozen Polycom IP501s into an existing customer for $2500. I made a few bucks and the customer is proud of the high tech instrument (and great speaker phone) on his desk. Why would you, as a vendor, want anything else? Also, they *love* the buddy list with presence. Mike P.S. Installing phones into a hotel is a different matter. Alexander Lopez wrote: Look at the Cortelco product line they have a hospitality (hotel/motel/Holiday Inn) offering that has CID, Speaker Phone, and is line powered. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thczv F. Thczv Sent: Friday, March 10, 2006 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Analog Desktop Phone I am looking for a good analog desktop phone to use with asterisk and my sipura ATAs. I know I want Caller ID, MWI, a few programmable buttons (for asterisk features), and no external power supply (so my users can dial 911 through the SPA-3000 when the power is out). I spent some time looking at the phones at Fry's today, without finding exactly what I need. Do any of you have any experience with this? Thanks, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 power over ethernet
As I understand 802.3af, the phones go through a negotiation with the unit supplying the power. I don't think it's a matter of -48VDC on a particular pair. I remember a schematic from years ago--it had each of the receive pair and the transmit pair going into a transformer winding, and that winding had a center tap for PoE. This is not something that *I* am going to screw with. The IP501 telephone set is the same for both PoE and local power. With the PoE cable, the 802.3af electronics (the negotiator) is a plastic thing in the cable. For the local power, there is a plastic thingie toward the wall end of the cable, and you plug the wall wart into the plastic thingie. Notice the advanced technical jargon here With local power, there is still only one cable one the desk--the power plugs into the cable towards the wall. Except for a power interruption, this has all the advantages of PoE. William M Conlon wrote: I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were powered over ethernet. Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power injector cable, they work with the switch in either its powered or unpowered ports. Anyhow, I hadn't seen any mention of how people power these phones, as I had planned on centralizing phone power on a UPS to supply my Asterisk server and POE switch. Now the question is: Can the Polycom AC-powered injector be used with a standard ethernet patch cable: switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: Polycom 501 which would allow me to power the Polycom AC adapters by my UPS. Or do I need to provide a UPS at each phone and run the ethernet like switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: Polycom 501 thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 power over ethernet
The IP501 does not have a power jack. You'll need one of the Polycom cables. William M Conlon wrote: My recollection of the marketing fluff was that we would just use our legacy network (cables) and the devices at both ends would figure out whether they were sourcing, sinking, or neither. In the case of the 501, it's the special Polycom cable, either with or without provision for an AC power adapter, that powers the phone. That's what I meant by saying the '501' itself is not compliant with 802.3af -- it needs a separate thingamajig [tech jargon :)]to be powered. Anyway I had hoped that I could just plug a CAT-5 patch cable from my RJ45 wall outlet into the phone. On Mar 5, 2006, at 5:17 PM, Michael Welter wrote: As I understand 802.3af, the phones go through a negotiation with the unit supplying the power. I don't think it's a matter of -48VDC on a particular pair. I remember a schematic from years ago--it had each of the receive pair and the transmit pair going into a transformer winding, and that winding had a center tap for PoE. This is not something that *I* am going to screw with. The IP501 telephone set is the same for both PoE and local power. With the PoE cable, the 802.3af electronics (the negotiator) is a plastic thing in the cable. For the local power, there is a plastic thingie toward the wall end of the cable, and you plug the wall wart into the plastic thingie. Notice the advanced technical jargon here With local power, there is still only one cable one the desk--the power plugs into the cable towards the wall. Except for a power interruption, this has all the advantages of PoE. William M Conlon wrote: I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were powered over ethernet. Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power injector cable, they work with the switch in either its powered or unpowered ports. Anyhow, I hadn't seen any mention of how people power these phones, as I had planned on centralizing phone power on a UPS to supply my Asterisk server and POE switch. Now the question is: Can the Polycom AC-powered injector be used with a standard ethernet patch cable: switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: Polycom 501 which would allow me to power the Polycom AC adapters by my UPS. Or do I need to provide a UPS at each phone and run the ethernet like switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: Polycom 501 thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk
[Asterisk-Users] Multi node call center
Are there any issues with having call queues and agents spread across two (or more) Asterisk systems? We would have a trunked IAX connection between systems, and the agents would use SIP phones. All agents are multilingual. One system would be on the West coast and another in Europe? Agents from different time zones signing on in their morning and signing off in their evening. The call center itself running 24/7. Issues? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Doug Lytle wrote: Michael Welter wrote: I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. RX Gains too high IRQ sharing of the of the ZAP device There is no ZAP device (it is a SIP-only implementation) and there are no interrupts being shared. High load of the machine The machine is totally idle. The T1 vendor noticed 2% packet loss during a ping flood originating from outside. We changed the Cisco IAD, and there is no longer packet loss, but we still have the clipping. Asterisk 1.2.4. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Doug Lytle wrote: Michael Welter wrote: Doug Lytle wrote: Michael Welter wrote: The machine is totally idle. The T1 vendor noticed 2% packet loss during a ping flood originating from outside. We changed the Cisco IAD, and there is no longer packet I've noted from employees that the volumes levels on the phones themselves, when set too high will cause crackling. Does the crackling coincide with talking on the local side? What firmware are you running on the Polycoms? I'm not on site, but I remember 1.6.4. It's not really crackling or popping that's the problem. The problem is with dropouts. It also seems that the trailing edge of each word will sometimes be lost (possibly a dropout). If you're familiar with the WWV time signal (303-499-7111), for the first 45 minutes of each hour there is a tone interrupted by a click every second (during the last 15 minutes it's just the clicks). When I listen to this on the Asterisk system, the tone only lasts for a fraction of a second and then silence until the next click. Is the phone (or Asterisk) performing echo suppression that drops the last part of the tone? Also, there are no ZAP cards in the system. What timing source does SIP use to play the incoming media stream? Thanks for your comments, Doug. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Doug Lytle wrote: I think the only time you need a timing source is if you are mixing audio streams, i.e. meetme, MOH. In which case you'd probably need to run ztdummy. Yes , ztdummy is running. I'm going to (temporarily) put a TDM card in the system just to eliminate that possibility. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call quality problems
I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. The WAN comes in from the Cisco IAD and into a LAN switch (DLink DGS-1005D w/ 802.1p) where the two public IPs are switched to different devices. One device is a FireBox device controlling a separate LAN with VPNs. The other device is eth0 on the Asterisk system. On the Asterisk eth1 is a 3Com 2226 LAN switch which connects Polycom IP501 phones. There are no PCs on this voice LAN. All ports on all LAN switches indicate full duplex. The quality problem doesn't appear to be volume related (a single call still has problems). The Polycom IP501s use SIP to the PBX, and the PBX uses SIP to the provider. The normal WWV time signal consists of a constant tone that is interrupted every second by a click. On the Polycom, each click can be heard, the tone starts, but the tone is clipped and there is silence until the next click. I've verified that QoS is enabled in the IAD. I would appreciate your thoughts. Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GPS-enabled cell phone/PDA
I would like to capture the lat/lon coordinates from a GPS-enabled cell phone or PDA. Is this possible? Must I subscribe to this information from the cellphone network provider, or can I capture it without charge? What devices will broadcast the coordinates? Is there a device that will broadcast its position inband that can be captured by Asterisk? Can an SMS message include coordinates? The subject will willingly carry the device and will be aware that his location is being monitored, so privacy rights are not an issue. The subject will make periodic calls to the Asterisk server in order to record his movements. Does anyone have experience in this area? Thanks, Mike -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk in SPA9000?
Andres wrote: Did Linksys really use Asterisk for the SPA9000 software? I certainly hope so. Have you checked what SNOM uses for their phones? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream web configuration utility
The GS2000 web interface won't work with Firefox. I had to use IE :-( in order to config my phones. Ron Bulthuis wrote: I just purchased a Grandstream gxp-2000, budgetone102 and a HT-386. Browsing to each device by IP address, I can get logged in using admin and I can see the advanced settings, however, if I try to change the settings and clicking the Change button, it just brings me back to ask for the password again.. I can't get into the Status page or any of the Account1-4 pages either. It just keeps bringing me back to the password screen. If I enter a bogus password it will tell me the password is incorrect so that seems to be working. I am using [EMAIL PROTECTED] v2.2 and the gxp2000 is using default firmware of 1.0.1.9 but it shouldn't be even looking at Asterisk yet. Do I need something more just to browse to these configuration pages in the device? All 3 units are doing the same thing. (I did not find anything in the FAQ's or documentation.) Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attack dialing
I think attack dialing means to dial all 10,000 number in an exchange, looking for modems and fax machines. BTW, Colorado Springs, Colorado has made it illegal to dial a number without intending to have a conversation sigh Probably something to do with NORAD or Space Command. Eric Bishop wrote: Anyone have eny elegant dial plan config for attack dialing? Basically I just want to automatically and continuously dial a busy until it is answered and then hand it over to a SIP hanset. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Toll-free number on a PRI
I have a toll-free number that is mapped to the main number of my PRI. When a call arrives, the called number is the main number, not the toll-free number. The PRI vendor is ICG, and they're saying the number gets mapped to the main number. I'm saying I want to see the toll-free number. Can they do this? What happens when I get a second toll-free number for a different business--will I be able to differentiate the called number? Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Switch for VOIP Applications
Ok, what's the best VoIP switch with PoE? Does anyone have experience with the D-Link DES-1526? Wiley Siler wrote: What is your port density requirement? For 24 ports the LinkSys SRW2024 is awesome. They street for less than $500 and have good QoS. For a smaller switch, they make a 12 port variant. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of calvis Sent: Monday, December 05, 2005 3:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Best Switch for VOIP Applications I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF errors
I'm getting the following messages when a call is answered by a SIP device: Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP Transmission error to 192.168.1.254:19262: Operation not permitted For a Cisco 7940 line, I have the following sip.conf entry: [desk2] type=friend username=desk2 secret=xxx host=dynamic dtmfmode=rfc2833 context=international canreinvite=no callerid=xxx3034144980 [EMAIL PROTECTED] nat=yes qualify=yes accountcode=xxx disallow=all allow=ulaw allow=g729 The Asterisk system faces the Internet on a public IP. The phone is behind NAT. Asterisk version is 1.0.7. What can I do to fix this problem? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP send errors
Michael Welter wrote: I'm getting the following messages when a call is answered by a SIP device: Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP Transmission error to 192.168.1.254:19262: Operation not permitted For a Cisco 7940 line, I have the following sip.conf entry: [desk2] type=friend username=desk2 secret=xxx host=dynamic dtmfmode=rfc2833 context=international canreinvite=no callerid=xxx3034144980 [EMAIL PROTECTED] nat=yes qualify=yes accountcode=xxx disallow=all allow=ulaw allow=g729 The Asterisk system faces the Internet on a public IP. The phone is behind NAT. Asterisk version is 1.0.7. It has nothing to do with DTMF. It is getting a few rejected rtp frames from the kernel 'sendto' function immediately after the call is answered. Does anyone offer some insight? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP send errors
Michael Welter wrote: Michael Welter wrote: I'm getting the following messages when a call is answered by a SIP device: Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP Transmission error to 192.168.1.254:19262: Operation not permitted For a Cisco 7940 line, I have the following sip.conf entry: [desk2] type=friend username=desk2 secret=xxx host=dynamic dtmfmode=rfc2833 context=international canreinvite=no callerid=xxx3034144980 [EMAIL PROTECTED] nat=yes qualify=yes accountcode=xxx disallow=all allow=ulaw allow=g729 The Asterisk system faces the Internet on a public IP. The phone is behind NAT. Asterisk version is 1.0.7. It has nothing to do with DTMF. It is getting a few rejected rtp frames from the kernel 'sendto' function immediately after the call is answered. Does anyone offer some insight? After answer, a few rtp frames are being sent from sip_write to the NATed address of the phone (192.168.1.43) and are being rejected. After that, rtp frames are correctly sent to the public address of the phone's firewall, and the conversation is normal. Can anyone offer some insight? Do I need to move to asterisk-1.2 before I go any further? Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Where to buy a T1 crossover cable for * and channel bank
Richard Reina wrote: Does anyone know where I can buy a 50ft crossover cable to connect my digium card -- I believe it's a T100P -- to my Adit 600. The one I have now works fine but I need a longer one. Two RJ45 plugs and a length of Cat3 cable (two pairs). Pins 12 on the first plug go to pins 45 on the second plug. Pins 12 on the second plug to pins 45 on the first plug. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] VOIPJET - are they down
Joseph wrote: On Fri, 2005-11-18 at 11:40 -0800, Luki wrote: Can anybody confirm if there is a problem with their server. The east cost server I use (64.34.45.100) works fine. --Luki That is strange, I can not make a out either through Teliax or VOIPJET. Calls via FWD are working fine. Do a traceroute on all three. Look for common failure points. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] PRI HDLC abort on dchan
Kevin P. Fleming wrote: Michael Welter wrote: Nov 15 20:09:15 NOTICE[27290]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 I ran top during that time, and there was no significant cpu usage. Probably interrupt starvation... are there any interrupts being shared, or does your NIC driver take an especially long time to handle interrupts? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you, Kevin. Removing the ohci_hcd modules on 177 solved the problem. [EMAIL PROTECTED] ~]# cat /proc/interrupts CPU0 0: 3775022330IO-APIC-edge timer 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 14: 24IO-APIC-edge ide0 169:5091470 IO-APIC-level libata, 3w- 177: 37749599 IO-APIC-level ohci_hcd, ohci_hcd, eth0 185: 101487948 IO-APIC-level eth1 193: 3774814937 IO-APIC-level t1xxp NMI:2726825 LOC: 3774551440 ERR: 0 MIS: 0 However: eth0 isn't plugged into anything. eth0 is assigned 192.168.2.4 while eth1 is attached to the Internet with a public IP. Both eth0 and eth1 are integrated into the motherboard. There should be no activity on eth0, so I don't understand why eth0 is getting interrupts and why there is a problem. Here is lspci for bus #1: 01:00.0 USB Controller: Advanced Micro Devices [AMD] AMD-8111 USB (rev 0b) (prog-if 10 [OHCI]) Subsystem: Advanced Micro Devices [AMD] AMD-8111 USB Flags: medium devsel, IRQ 177 Memory at feafb000 (32-bit, non-prefetchable) [size=4K] 01:00.1 USB Controller: Advanced Micro Devices [AMD] AMD-8111 USB (rev 0b) (prog-if 10 [OHCI]) Subsystem: Advanced Micro Devices [AMD] AMD-8111 USB Flags: medium devsel, IRQ 177 Memory at feafc000 (32-bit, non-prefetchable) [size=4K] 01:00.2 USB Controller: Advanced Micro Devices [AMD]: Unknown device 7463 (rev 02) (prog-if 20 [EHCI]) Subsystem: Advanced Micro Devices [AMD]: Unknown device 7463 Flags: medium devsel, IRQ 177 Memory at feafe400 (32-bit, non-prefetchable) [size=256] Memory at feafe000 (32-bit, non-prefetchable) [size=32] Capabilities: [80] Debug port Capabilities: [88] Power Management version 2 01:07.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 6159:0001 Flags: bus master, medium devsel, latency 64, IRQ 193 I/O ports at b800 [size=256] Memory at feafd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 01:0a.0 RAID bus controller: 3ware Inc 3ware Inc 3ware 7xxx/8xxx-series PATA/SATA-RAID (rev 01) Subsystem: 3ware Inc 3ware Inc 3ware 7xxx/8xxx-series PATA/SATA-RAID Flags: bus master, 66Mhz, medium devsel, latency 64, IRQ 169 I/O ports at bc00 [size=16] Memory at feafe800 (32-bit, non-prefetchable) [size=16] Memory at fe00 (32-bit, non-prefetchable) [size=8M] Expansion ROM at fea8 [disabled] [size=64K] Capabilities: [40] Power Management version 1 01:0b.0 VGA compatible controller: ATI Technologies Inc Rage XL (rev 27) (prog-if 00 [VGA]) Subsystem: ATI Technologies Inc Rage XL Flags: bus master, stepping, medium devsel, latency 64, IRQ 11 Memory at fd00 (32-bit, non-prefetchable) [size=16M] I/O ports at c800 [size=256] Memory at feaff000 (32-bit, non-prefetchable) [size=4K] Expansion ROM at feac [disabled] [size=128K] Capabilities: [5c] Power Management version 2 01:0c.0 RAID bus controller: Silicon Image, Inc. (formerly CMD Technology Inc) SiI 3114 [SATALink/SATARaid] Serial ATA Controller (rev 02) Subsystem: Silicon Image, Inc. (formerly CMD Technology Inc) SiI 3114 SATARaid Controller Flags: bus master, 66Mhz, medium devsel, latency 64, IRQ 169 I/O ports at cc00 [size=8] I/O ports at c480 [size=4] I/O ports at c400 [size=8] I/O ports at c080 [size=4] I/O ports at c000 [size=16] Memory at feafec00 (32-bit, non-prefetchable) [size=1K] Expansion ROM at fea0 [disabled] [size=512K] Capabilities: [60] Power Management version 2 01:0d.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5705 Gigabit Ethernet (rev 03) Subsystem: Broadcom Corporation: Unknown device 1644 Flags: bus master, 66Mhz, medium devsel, latency 64, IRQ 177 Memory at fea9 (64-bit, non-prefetchable) [size=64K] Expansion ROM at fe9e [disabled] [size=128K] Capabilities: [48] Power Management version 2 Capabilities: [50] Vital Product Data
[Asterisk-Users] PRI HDLC abort on dchan
I used sftp to move the RC2 files to my * server. During the whole time of the transfer, asterisk continuously reported: Nov 15 20:09:15 NOTICE[27290]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 I ran top during that time, and there was no significant cpu usage. The system uses Asterisk 1.0.7. The cpu is an Opteron on a Tyan motherboard. Can anyone explain this? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Notices at beginning of call
I get 8 to 12 of these messages at the beginning of a call to a Cisco 7940 SIP phone: Nov 13 17:27:56 NOTICE[27203]: rtp.c:1146 ast_rtp_raw_write: RTP Transmission error to 192.168.1.254:32472: Operation not permitted Other than these messages, the call proceeds normally. From the code I see Asterisk is attempting to send DTMF digits. In SIPDefault.cnf I have dtmf_inband: 1 In sip.conf I have tried dtmfmode=rfc2833, dtmfmode=inband, and dtmfmode=info. Asterisk 1.0.7. What is confusing is that these messages started appearing this past week with no apparent changes to the configs. Can anyone help me with this? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Registration from Verizon DSL
I have a client who is unable to register her SJPhone on my Asterisk server. She is using a Westell DSL router connected to Verizon. Others in her group, using cable modems, are able to register. The group is located in the Dallas area. Is Verizon still blocking SIP registrations? Is there something about the Westell that needs to be changed? From what the client says, outbound traffic is unlimited. In sip.conf, I have nat=yes and qualify=yes. Thanks -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX channel options
I have an installation with four Qwest POTS lines. For some unknown
reason, Qwest drops the first digit in the dial string, and the call
fails. To fix that problem, I put a 'W' in the dial string:
QWEST=Zap/g2
exten = _9303NXX,1,Dial(${QWEST}/W${EXTEN:1})
The client has since complained that, when all four lines are busy, he
cannot make a local call. So I provided the ability to roll over to
another system to complete the call:
TELECOMMATTERS=IAX2/[EMAIL PROTECTED]
exten = _9303NXX,1,ChanIsAvail(${QWEST}${TELECOMMATTERS})
exten = _9303NXX,2,Cut(MYCHANNEL=AVAILCHAN,,1)
exten = _9303NXX,3,Dial(${MYCHANNEL}/W${EXTEN:1})
exten = _9303NXX,4,Hangup
exten = _9303NXX,102,Congestion
This works fine for the Qwest line, but Asterisk doesn't absorb the 'W'
for the IAX call--the 'W' is sent as part of the dial string.
Is there a solution for this?
TIA
___
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Re: [Asterisk-Users] Adit 3104 configuration
I just installed several 3104s in S. Calif. Didn't have any problems--I was able to call from one line to another on the same unit and between lines on different units. Jerry Jones wrote: Has anyone been able to get the 3104 to register more than one line correctly? It seems to work OK for the first line, but as soon as I turn on more than one it appears that only the last one is actually registering corectly. The 3104 sometimes indicates the line is registered, but * says not. This looks like a very useful unit and would really like to get it to work. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] RxFax/TxFax - Compile Problem
What version of libtiff are you using. Has anyone tried 3.7.x with spandsp? Doug Lytle wrote: Alexander Lopez wrote: I have used the pre20 package, with the latest CVS-head. COmpile goes cleanly, NO ERRORS. then I get this when I try to load asterisk -cvv [app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Sep 20 14:00:23 WARNING[5924]: loader.c:554 load_modules: Loading module app_rxfax.so failed! No matter what I do it compiles clean but errors out with undefined symbol errors. Does anyone have a clue on this I'm having the same issues, so I've installed Asterisk on my laptop, did a fresh compile of libtiff and spandsp pre2-20 and started Asterisk. Asterisk app_txfax and app_rxfax compile without issues and Asterisk starts without complaining I'm going to remove and re-compile spandsp and libtiff tonight to see if it makes any difference with the effected machine or not. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SUCCESS - 512 Simultaneous Calls with Digital Recording
Patrick wrote: On Tue, 2005-09-20 at 18:37 -0400, Matt Roth wrote: List users, Over the last few days we have been working with MCI's development lab to test our Asterisk setup. We were using a piece of hardware called an Abacus 5000 that is capable of creating and terminating thousands of SIP calls. Initially, we could not get past 64 simultaneous digitally recorded calls without having call quality issues including dropped calls. We identified an I/O bottleneck and rectified it by digitally recording to a RAM disk. Using this method, we were able to digitally record 512 simultaneous SIP-to-SIP calls with 100% call completion. Our plan is to use the MONITOR_EXEC hook to call a custom program that will copy files to the hard disk at call completion. Matt, Interesting stuff. Did you test copying of the recordings on the ramdisk to a local harddisk also during that 512 call load? Just wondering if that copy action wouldn't also create an I/O bottleneck and cause call quality issues under load. Did you consider using remote storage e.g. via nfs, a fibre channel or iSCSI link to a SAN? I use 3Ware RAID cards in my systems with write cache turned on. I'm wondering if this presents a reduced interrupt load on the system than directly-attached hard drives? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] unlocking cisco 7940 phone
I got five of these doctor's phones as well. You need your server set-up with dhcp and tftp. In the tftp directory, you'll need files SIPmac.cnf and SIPDefault.cnf available. You'll also need the current SIP firmware load. Then, you have to let the phone power up, then enter the password 'cisco', the go to settings-3-28 (remove config) and also -25 to enable DHCP. Then hit 'yes'. If your files are ok then the phone will reboot, upgrade its firmware, and read the new configuration. I performed this procedure five times this morning and it worked every time! 5¢ please :-) Rich Adamson wrote: Unfortunately, it didn't work. Do you know any other ways? Brian S. Reale President Colosa Inc. Tel: 305.675.1400 ext. 201 Fax: 305.402.0282 www.colosa.com www.ProcessMaker.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Saturday, September 17, 2005 9:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] unlocking cisco 7940 phone try going into settings and then unlock config and try typing in cisco On 9/17/05, Brian [EMAIL PROTECTED] wrote: I have a Cisco 7940 phone with a locked SIP configuration. There is no tftp server configured in network settings. Does anyone know how to get this phone to upgrade its firmware and thereby unlock the SIP settings? (The **# combination on the Cisco manuals does not work). -brian I just had to do one the other day; the phone shipped with mgcp installed with some Doctors name appearing on the screen. It had been locked down pretty tight. The only way that I could do it is to use a sniffer (eg, ethereal) to see the IP addresses it was trying to reach. I then put the phone on a network that had a linux server with that exact addressd on it. The phone happily read the tftp files that I had prestaged on the linux server, and upgraded itself to the sip image. The upgrade process wiped out all passwords, etc, and reset the phone to factory defaults for sip (except for the password, which I specified in a tftp config file). Once that was done, I could access the phone menues just fine and reset everything to the way that I wanted it. If you're not comfortable reading packet traces, then you'll probably have to send the phone to someone that is. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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-Email for Hosted PBX
I'm proposing to install an Asterisk PBX at a collocation facility for a remote customer. Each of the customer locations will have an SPA-3000 with the FXO port connecting a POTS circuit and the FXS port connecting a fax machine or red phone. In addition to voice traffic, the customer has a high volume of incoming and outgoing faxes. Would it be possible, using g711 between the SPA-3000 and server, to have spandsp/rxfax receive a fax from the POTS circuit via the SPA-3000? From the locally attached fax machine? (I realize that packet loss will have a adverse effect on fax transmissions.) Would I be better-off attaching the fax machines to a Mediatrix 2102? Any help is appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P stops answering
Same problem here. ./asterisk stop;./zaptel restart;./asterisk start seems to get it working. Question: will ztcfg -vv alone get it working? Andy Howell wrote: I have a weird problem in which my digium card stops answering. After running for a couple days, incoming calls are not seen. Running asterisk -r shows no incoming calls. Restarting Asterisk does not help. After a reboot it is fine. Any ideas? Thanks, Andy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on AMD64
I have several installations running Asterisk over FC3 x86_64 on Tyan Opteron motherboards. I also have one installation on an Athlon 64. When compiling zaptel, be sure to 'make linux26' for the 2.6 kernel. Massimo De Nadal wrote: Joseph ha scritto: On Mon, 2005-09-12 at 09:56 +0200, Domjan Attila wrote: On Mon, 2005-09-12 at 08:51 +0200, Dave Cotton wrote: On Sun, 2005-09-11 at 20:58 -0600, Joseph wrote: Does anybody runs Asterisk on AMD64? I can compile it on Gentoo, and start Asterisk a command line but as soon as I connect any device (like Sipura ATA ), asterisk crashes. Runs well here. here too :) What version are you running? I think you can't run asterisk on a 64 bit linux version. Certainly it works well on a 64 bit amd processor, but in x86 mode... Am i right guys ?? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Hotel Setup?
Have you seen the 3Com LAN-switch-in-a-wall-jack device? It's a four port device that could also be used for the guest's PC. It can supply PoE to the phone using a wall wart or PoE on the incoming LAN circuit. You'll need some logic between Asterisk's management interace and the property management system's (PMS) telephone interface. The PMS will expect call detail to be received in real time. Depending on the PMS, outside calling is switched on during checkin and switched off upon checkout. Some PMS restrict toll calls if non-adults occupy the room. Also, some PMS have certain codes that the maids (and supervisors) dial from the room to indicate availability. And the staff need to be aware of any 9-1-1 calls. You'll have to be able to empty a room's voice mailbox upon checkin. And some properties want the guest's name in the caller id. So it's not simply putting a SIP phone in the room... [EMAIL PROTECTED] wrote: Small world. The Inn was going to work on absorbing the cost of the system and the VoIP service. The phones would be just cheapie grandstream phones, which work out to about the same as regular analog phones. More features, no cost, the owners are thinking they can lever this edge to attract more business customers and such. I have a serious problem with more hardware in each room. More things to get stolen, broken, etc. Plus have the costs of the adapter, plus the cost of the phone, and you are right back to a Budgetone price. More cables, more things to play with. Plus you need to provide an outlet, whereas I could do PoE and eliminate yet another source of problems. I would like a phone, and a cat5 cable. That's it. You want internet access, use wireless or the cable. You can't depend on the customer to do the right thing. Ever. In fact, count on them doing the wrong thing. All the time :-) On a sidenote, I would like to know how using a port based PVLAN setup and DHCP won't provide adequate isolation between rooms. Am I unclear on something? ~kurth ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] g729 test
Michael Welter wrote: My preferred LD vendor requires g729 and SIP. Is there a method to test, prior to initiating a call, whether a g729 codec is available? Will ChanIsAvail test g729 availability? To clarify: I have n g729 licenses for my system. If I have n g729 calls in process then I don't want to attempt another g729 call. Is there a method to test whether a g729 codec license is available? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 test
My preferred LD vendor requires g729 and SIP. Is there a method to test, prior to initiating a call, whether a g729 codec is available? Will ChanIsAvail test g729 availability? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Good Polycom Dealer?
Maybe you should read tritechcoa's return policies before you start recommending them. After suffering through their RMA juggernaut, I'll never again do business with them. Tarpo, Louie wrote: We use both voipsupply and tritechcoa. We've had no problems with either one. I've received firmware from each company. Louie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kenny Kant Sent: Tuesday, September 06, 2005 3:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Good Polycom Dealer? Could any of you provide me information on a good Polycom phone dealers to utilize. One who provides firmwares ..etc Thank you! Kenny __ Click here to donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] kernel panic
I've just loaded zaptel 1.0.9 on a new 2.6.12 system (FC4 with updates). The system has a TE110P card, and zaptel.conf is configured for an E1. When I do a 'zaptel stop' I get a kernel panic. Has anyone else seen this? Thanks, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] required packages for asterisk on FC3/FC4
I'm going to build an Asterisk system on a 1G Compact Flash card with NFS mounts for /var/spool, /var/log, etc. Does anyone have information on which packages are required for the CF card? Also, I would like to set the CF card to read only. Does anyone have information on which directories are written to during Asterisk's normal operation? For example, I see the file asterisk.pid in /var/run. Thanks Damon Estep wrote: Can anyone shed some light on which of these packages are required and what component requires them? I am in the habit of putting them on, but in a few cases am not sure if they are still (or were ever) needed. qt-devel rpm-build gcc gcc-c++ redhat-rpm-config gtk2-devel y ncurses-devel readline-devel bison krb5-devel openssl-devel cvs patchutils libidn-devel kernel-smp-devel (smp machine) flex (for a particular brand of pri card) Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HDLC on T1
I have a new client who is about to order an Integrated PRI T1--eight voice channels and the remainder data. He's not sure whether Qwest will provide a router to split the voice and data. I remember we had a lot of trouble with HDLC in Linux kernels past 2.4.19. Have those problems been fixed in the 2.6 kernel? Will I be able to split the voice and data channels using zaptel? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Motherboards and IRQs
Paul wrote: jennyw wrote: Someone mentioned earlier (I can't find the message now) that they had a motherboard that allowed you to change IRQ assignments in BIOS. Does anyone happen to know how to identify motherboards that can do this? I'm going to put together a new machine now and I'm having trouble picking a motherboard for it (ordering from Dell or other online vendor is not an option, since I need this in the next couple days). This is to provide another avenue for avoiding IRQ conflicts with the Digium TDM400Ps. Thanks! Jen intel 875p chipset and ECC RAM works well. I have built several servers with intel and abit boards using this chipset Always use a good UPS rather than a surge protector outlet strip. You can get an APC brand with USB signalling for as little as $50. It works with linux. That means the linux server will see when the battery is low, prepare itself for shutdown and then tell the APC to kill its output. When power returns the APC will turn outpput back on and the linux server boots gracefully. When buying CPU I usually go with the retail boxed instead of OEM. It doesn't cost much more. It includes the cooler along with a good warranty. If it fails you get a new cooler along with the new chip. Obviously they can't try to get out of warranty replacement based on the cooler selection if it was their choice in the first place. I stopped using athlons when I could no longer get boards with the AMD chipset(supports ECC). You can probably still get boards designed for dual athlon. I asked AMD why and they told me the 760 chipset for single athlon was no longer in production. It seems the average consumer doesn't appreciate higher quality features enough. It's too bad because I always liked AMD stuff. When I needed fast reliable chips for some specialized processors I designed it was AMD that gave me the first samples and they always gave the purchasing people good pricing for production needs. I'm only mentioning this because there has been some banter about AMD vs. Intel here. If the right motherboards were available for single socket 7 servers I would have stayed with them. Their service is excellent. They give me free overnight shipping and 2 weeks to send in the defective part. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Try the Tyan S2850--single Opteron. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] asterick and festival...Help!
John Gruber wrote:
Earlier this afternoon I had this working
exten = 2890,1,Answer
exten = 2890,2,GoTo(12)
exten = 2890,12,Wait(1)
exten = 2890,13,Festival('I can say numbers like')
exten = 2890,14,SayNumber(1230001,f)
exten = 2890,15,Wait(1)
exten = 2890,16,HangUp
I was so very proud of myself...
All of a sudden after a reboot I get the following from the same
call plan
--- (9 headers 0 lines)---
-- Executing Festival(SIP/1000-2915, I can say numbers like) in
new stack
== Parsing '/etc/asterisk/festival.conf': Found
== Spawn extension (mytest, 2890, 13) exited non-zero on 'SIP/1000-2915'
and of course the call exits.
Here is my /etc/asterick/festival.conf
[general]
host=127.0.0.1
port=1314
usecache=no
cachedir=/var/lib/asterisk/festivalcache/
festivalcommand=(tts_textasterisk %s 'file)(quit)\n
Everything is running on the same box. I have rebooted... nothing is
var log messages either.
The local festival_client connects and I can put in (SayText I can say
numbers like) and it works great.
The festival_server log show only this for the calls from asterick:
client(11) Thu Aug 18 17:53:01 2005 : accepted from (my machine name here)
client(11) Thu Aug 18 17:53:01 2005 : disconnected
So it looks like it is connecting right.
Delete the files in /var/lib/asterisk/festivalcache and then try it
again--see if the behavior changes.
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Re: [Asterisk-Users] Festival error
festival_server.scm -- should be in /usr/share/festival grep for '(localhost) and replace with nil (don't use double quotes). Innocent Evil wrote: Hello, I have installed festival (the rpm package came with fc4). But getting this: client(1) : rejected from myserver not in access list whenever I try to access it from asterisk . I found in documentation: If you see a message such as: client(1) : rejected from myserver.mydomain.com not in access list then edit the festival/bin/festival_server startup script to include that FQDM in the line with localhost.*. but I dont see any file named 'festival_server' in my fc4 box. How can I get arround this issue? Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Problem
I'm attempting to use Festival with Asterisk on an x86_64 system. This IVR application works ok on a P4 system. I'm using the FC3 x86_64 distro on a single processor Opteron system. Festival by itself (using the command line and speakers) seems to work ok, and Asterisk without Festival works ok. When the Asterisk dialplan calls Festival, however, Festival reports a disconnect and Asterisk's Festival command does not complete. Later, when I shut down the system for reboot, I get a kernel panic. I've tried both the FC4 Festival rpm as well as the source download from festvox.org. I modify the siteinit.scm file as per the wiki page, and I use the stock festival.conf file in Asterisk. Has anyone experienced this behavior, and is there a workaround? Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Nortel Option 11 and TE110P of Digium
Alvaro, have you considered putting the Asterisk box between the PSTN and the Nortel. In the initial Asterisk configuration, all you do is hand-off incoming calls to the Nortel. When someone on a Nortel set dials an outside call, all the Asterisk box does is hand it off to the PSTN. In this configuration the Nortel configuration does not have to be changed (it doesn't know the configuration changed). Also, you can then make incremental changed to the Asterisk box (like adding some SIP telephones). I have an account here in Denver, Colorado where we have done what I described (Nortel/Asterisk). Contact me off list if you would like more information. Michael Welter +1-303-718-2804 Paul Belanger wrote: See inline for clocking sources: Alvaro Parres wrote: Well the Actual Digramm of conecctions is: E1 E1 PSTN -- NORTEL - ASTERISK Signaling pri_network -- pri_cpe pri_network --- pri_cpe Local Lines Actually the client the only think want is the solution working even how much cost it. The proyect was start by a employ and hi never can do this.. Thats why i'm asking to see if i take the proyect o not. So with the PRI it's going to be easy all the work ?? Only one question the Nortel guys here, say that they need one more clock to have a PRI card, is this correct On 8/5/05, Paul Belanger [EMAIL PROTECTED] wrote: Hello, See comments inline Alvaro Parres wrote: Hi list: I have a client that needs to connect a Asterisk PBX with a TE110P of Digium and one Nortel Option 11. Actually the Nortel Option 11 have a AMI E1 card. With it the have problems of clock sync. Is the Nortel the CPE or Network side? They can change the AMI CARD to a PRI CARD, te questions are: 1) Which model of PRI is suggest for this ? NI2 or DMS100 2) Some one have already do this ? Yup, have a site up with a TE405P. 3) Is there form of correct de AMI problem ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need Advice
I need to place a SIP FXO gateway in Central America. I've been looking at Quintum products, but the prices are about $150/FXO port. I have a Dell SC400 on the shelf, and I'm considering just installing Asterisk and two TDM04B cards and shipping it down. Does anyone have multiple TDM cards in the SC400? FXO ports on a TDM card are about $75/FXO port. With the Dell, there will be the advantage of trunked IAX (we're paying for bandwidth on both ends). Can anyone tell me if I'm missing something. The mains are 110V. What about answer and disconnect supervision (I'm assuming there isn't any on the FXO circuits). Is there some other way to do this for around $500? Thanks for your help, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Dell Hardware
[EMAIL PROTECTED] wrote: Mmhh nice !! So, why did Digium forbid it :)? If Dell is so bad... why is a Dell 2850 server one of the two listed on the compatibility list for ABE? http://www.digium.com/index.php?menu=product_detailcategory=softwareproduct=ABEtab=compatibility Does the Dell 2850 have anything to do with the Tyan 2850 mother board (single Opteron processor)? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Mixed Voice/Data T1
Chris Mason (Lists) wrote: We have a server running Asterisk and shorewall, three network interfaces and a T1 card, it functions as our firewall, pbx and connects to an Adtran 600 for FXS/FXO. We currently have two internet feeds, hence the three NICs. Our 10 PSTN lines are currently delivered POTS to our FXO and rather lousy service, volume is low and voice quality is dull. The Telco is offering me a Fractional T1 for our data needs. I thought that the best way would be to have the voice and data delivered on the T1. How difficult would it be to take the T1 into our Sangoma T1 card and seperate out the voice to asterisk and the data as our internet feed? We have an Eschelon integrated T1, and they installed a Vina Integrator to split the voice and data. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Braodvoice - UK Non Geographic Numbers
Russell Horn wrote: Since May 05 I have been unable to call any non-geographic number in the UK via Broadvoice. Thse are numbers such as the 0800 range (free to call) 087xx (local / national rate calls). Broadvoice support have been unhelpful, and can't say if there's any intention to fix this. A case has been upen since May 24 without any updates. Is anyone else having this problem? Has anyone else spoken to broadvoice about it? Did you get any further? Is there any indication it might be resolved? The last customer rep I spoke to recommended I close my account if I need to dial these numbers - I'd prefer to keep my phone number, but if all else fails... Russell. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I lost a client because of this. BT will not allow premium numbers to be called from outside the UK. I even tried it from an ITSP in the Netherlands, and the call didn't go through :-( The ATT monopoly is gone. Hopefully, BT's time will come--the sooner the better. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 Channel Bank Recommendation
I will be installing an Asterisk system in Honduras, and I need to convert many 2-wire analog circuits to E1 PRI. I have no idea if there is answer or disconnect supervision on the POTS circuits. An E1 from Hondutel seems to be out of the question because they will only provide SS7 signaling (not PRI), and the SS7 signaling would be on an extenal V.35 circuit. Does someone in Latin America have experience with E1 channel banks, and would they be willing to share their experience? Gracias ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Telephoning Announcements -- Suggestions?
Scott Nelson wrote: In the subdivision where I live, we have a well that time to time has problems. How about just fix the well :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
