Re: [asterisk-users] ATA recommendation??
Hello, I want to ask that if thee are some ATA decives that i can use to connect mutliple analog phone lines to my VOIP system.. I mean for example an ATA device with 24 ports with 24 independent SIP accounts. For example for some dormitories in my area, i want to put an ATA device and move existing lines to VOIP accounts. Only problem is, if i dont give seperate SIP accounts for all ports, i can not control who is calling where... And the billing system will also be a problem in that case. These are called SIP Gateways. There are several manufacturers who make them. I would suggest Audiocodes, Vega, or Carrier Access as starting points. Yes they come in 24 and 48 port versions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom MWI.
On Mar 19, 2009, at 9:05 AM, Ken D'Ambrosio wrote: Hey, all. I'm all over MWI, but I gotta say that I think the Polycoms go a bit over the top. The blinking LED is enough for me; how do I disable the stuttered dialtone and the audible warble? I've looked through the config files, but there are a HELL of a lot of options, and I haven't been able to find those particular ones yet. Thanks! sound_effects patterns MISCELLANEOUS MESSAGE_WAITING se.pat.misc.1.inst.1.type=silence se.pat.misc.1.inst.2.type=silence se.pat.misc.1.inst.3.type=silence / /MISCELLANEOUS /patterns /sound_effects ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Credit Card processing machines
On Feb 17, 2009, at 1:20 PM, David Gibbons wrote: snip We will be testing the ADT connection heavily this week. The modem connections to my understanding are 2400 baud. Over G.711U and a T1 I don't see why this wouldn't be as solid as a POTS line, but our tests will tell! /snip We do *fax* in this way and it works like a charm. We can hit much more than 2400 baud I think too. --Dave Most alarm systems around here use bursts of dtmf - not an actual modem to communicate with alarm central. Yes I have seen these have many issues with voip in the path. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Gmail is broken (was: Re: WiFi SIP phone w/VPN?)
Sorry about off-topic, but can you advise the mail client who is able to organise the web mailing list topic as web interface does ? (i mean by blocks/topics) I wold be glad to use something else with the same usability, but dont see any alternative. Thank you Just turn on threading ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
You will have a hard time finding a 24 port POE without fans - too high of a power density. Do you really need 24 ports? perhaps a 12x12 otherwise multiple 8 or 12 port models may work Do let us know if you find a 24 port without fans. On Jan 31, 2009, at 2:06 PM, Claus Herwig wrote: Hello, I need to put a 24 port Gig PoE switch into a small office – no computer room / rack etc. All CAT5 terminates near the owners desk (smart huh?). I had a similar problem some days ago. 24-port GBit Switch in the middle of a classroom... I ended up with a kind of semi-loud setup: I bought a 3Com 3CBLSG switch (this is without PoE, but there is a PoE version of it). There are two quite noisy 40x40x10 fans inside. I replaced the two with one 40x40x20 ebmPapst silent fan (model 412/2, 18dB), left the other fan offline and mounted the whole thing vertically so that convection supports the remaining fan. I tried with two silent fans (enough space inside), but this still was too noisy. Some measurement indicates the cooling is sufficient this way. But understand that I've no long term data, as I installed this setup just two weeks ago. And of course your warranty is void ;-) Greets, Claus -- CHECON EDV-Consulting und Redaktion Claus Herwig * Barer Straße 70 * 80799 München +49 89 27826981 * Fax 27826982 * c.her...@checon.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
Instead you could always get a SIP/IAX provider. On Jan 27, 2009, at 11:56 AM, Jon Pounder wrote: Michael Higgins wrote: At least here in Canada - DSL just seems to have killed BRI - you practically have to know the secret handshake to even be allowed to provision one any more. It killed it as an internet transport which was its most widespread use, however its many benefits as a digital phone line are being largely ignored. I barked up the same tree you are barking for a while and just gave up - lots of you could buy this and try it, but no proven solution. Kind of expensive to get a line put in and buy hardware for a maybe. Years ago we had tons of BRI circuits around I could have tried this on, but thats long gone. Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have found. What I did find left me with the impression that USA 'BRI', uh, '2B1Q' protocol(?) is not supported by *any* hardware vendor, at all, period, nor is it tested and proved in the software... stack(?), in one related branch or another on the OS side. A couple of direct inquiries to card vendors have dead-ended with a flat no, or requests for development funds(!) -- apparently there is code for one card, one vendor, that runs against 'bristuff', or did at one time, but wasn't maintained through several Asterisk releases (if the code was even released to the community... IDK). Is this common, that someone codes to their chip on their card and sells it to one or two consumers, then lets it drop and never gives the code up for continued development? (It seems contrary to GNU/ Linux licensing conventions, but, again, I'm not paid as a software developer. I just think they might have sold more cards with a less proprietary approach.) Anyway, can I, with confidence, state (to the $employer) that Asterisk on linux via USA 'BRI' digital lines simply isn't possible? (In that, obviously, I can't pay for development nor do beta testing, each with vague hope that it might work okay someday...) If this is the case, then I must use multiple analog lines to access PSTN, or pay premium for 'PRI' pipes (80% of which we will never need)... is that about correct? Thanks in advance for any pointers, specific RTFM suggestions, any help appreciated. If there is a different list to post this query to, I'm not (yet) aware of it. Cheers, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Description of Zaptel/DAHDI E1 alarms
In dahdi_tool, there are three more indicators of error: IRQ misses Bipolar violation CRC error As I understand it now, these should be error counters and they provide additional information in case of RED alarm state. Actually you need not be in RED alarm to have these. Just know that any quantity is bad. Some (BPV/CRC) occur when line is connected but should stabilize and disappear immediately else there is a problem if they keep counting. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote RTP
On Jan 16, 2009, at 10:38 AM, Gabriel Ortiz Lour wrote: Hi all, Suposing that 2 SIP phone register at a remote (internet) asterisk, what is the best way, if any, to make the RTP traffic go phone to phone, whithout using the internet conection (asterisk)? Allow reinvite? Assuming both are not behind NAT. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ghost in the Channel-Banks
On Dec 22, 2008, at 10:38 PM, Martin Lima wrote: On Thursday 18 December 2008, Justin Phelps wrote: I've been struggling with an ongoing problem the last month. Here is the layout of the wiring: T1 from ISP DiTech Echo Cancel device Voice Channel-Bank (same) T1 from ISP (same) DiTech Echo Cancel device asterisk1 server zap card fax channel bank (same) T1 from ISP (same) DiTech Echo Cancel device asterisk1 server zap card asterisk2 server Now, let me explain the symptoms. d-channel errors on the asterisk1 server on span1 (which is the line coming from the echo cancel from the ISP). asterisk2 server isn't being used as far as I can tell. I've got a red alarm on the port on asterisk1 that asterisk2 is plugged into. I would bet your asterisk2 server was meant for some kind of transition to a different setup. Is there at least some dialplan inside? sip.conf? iax, voicemail etc...? Something does not fit here. If you have the T1 from the ISP going to the echo can, then it cannot go to more than one device. It is not a MUX as far as I remember, have used to great effect in past. Each T1 in matches up to a T1 out. faxes through the fax channel bank are working most of the time. There seems to be problems with multipage faxes. Not isolated to a particular port on the channel bank. As I understand it, echo cancel for faxes is a bad thing, so I don't understand why the previous IT Admin here setup the system as such. Some echo cancellers detect a fax transmission and turn echo cancellation off. In your case if I understand your setup correctly such a device is ambiguous at the best... Possibly it used to be another set of voice lines converted to fax without changes in configuration? I dont really understand the role of echo canceller on E1/T1... Guarantee the Ditech deteects fax and cancels echo can. However any IP in FAX path is problematic for most FAX. If possible turn off ECM. If symptoms are mainly on a single fax machine make sure you have enough memory to buffer entire fax. Some systems let you print as they come in, but I have also seen some which are still trying to buffer while printing and run over. This showed up with a mortgage company doing 50 page legal forms. First 20 or so were fine then started bombing. Best method I have found to troubleshooting FAX is to use a machine which generates a T30 trace output upon completion. Some voice phones on the voice channel banks were not recognizing tones Why the phone should recognize tones? It just generated them while dialing. when dialing. That seems to have been resolved after power cycling the channel bank a few times, and restarting asterisk2 (odd that there doesn't seem to be anything active on asterisk2) Looks like some strange ground loops in your wiring, power issue or something similar. I've been working with the ISP on the d-channel stuff, and things seem to get a little better as they reset equipment, but the d-channel errors have not gone away. Some as above... BTW they will always tell you they reset something no matter what they have really done... :-) Again if this is a PRI, how is signalling being done for 2 T1 connections when only one D Channel. I suspect a partial PRI? I really need advice on these problems. From what I've said, where do you think the problem lies? in the channel banks? in the echo canceller? in the asterisk2 or asterisk1 server? With the ISP? Hard to guess without deeper knowledge of your setup. Intermittent errors and hardware lockups are often caused by power conditions, potential differences and spikes in the powerline. (two pieces of equipment connected together but plugged into two different power outlets coming from two opposite ends of the building can cause real headache!) Check it first. Then you may want to continue what your former IT admin tried to start :-) Are D Channels mostly working just intermittant? I assume you have Ditech set for passthrough on these channels? ie no echo cancel? Martin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dedicated Fax Line
Simple. A PRI can easily have multiple trunk groups. They just assign chan 1-22 to trunk group 1. Chan 23 to trunk group 2. D to chan 24. As an example, adjust to suit your needs. On Dec 16, 2008, at 9:27 AM, Andrew Thomas wrote: I can only assume it's a T1 thing - as E1's tend not to have that facility. Oh well, you live and learn :) Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Tim Nelson -- Sent: 16 December 2008 15:08 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] Dedicated Fax Line -- -- I've worked with many providers who are able to do this. In fact, -- we're using such a setup on our office PRI. I'm not sure how they're -- achieving this on their end however... -- -- Tim Nelson -- Systems/Network Support -- Rockbochs Inc. -- (218)727-4332 x105 -- -- - Andrew Thomas a...@datavox.co.uk wrote: -- -- Since when can you segment PRI channels off at the telco end? I -- know -- you could do with DASS - but I'm not aware you can do it with PRI. -- -- -- Andrew Thomas -- Technical Services Manager -- DataVox Ltd -- Saddleworth Business Centre -- Huddersfield Road -- Delph, Oldham -- OL3 5DF -- -- ___ -- -- Bandwidth and Colocation Provided by http://www.api-digital.com -- -- -- asterisk-users mailing list -- To UNSUBSCRIBE or update options visit: -- http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone know which vulnerability specifically they are referring to?
http://www.networkworld.com/news/2008/120608-fbi-criminals-auto-dialing-with-hacked.html?Inform=nlnetht=rn_120808nladname=120808dailynewsamal Criminals are taking advantage of a bug in the Asterisk Internet telephony system that lets them pump out thousands of scam phone calls in an hour, the U.S. Federal Bureau of Investigation warned Friday. The FBI didn't say which versions of Asterisk were vulnerable to the bug, but it advised users to upgrade to the latest version of the software. Asterisk is an open-source product that lets users turn a Linux computer into a VoIP telephone exchange.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Oct 29, 2008, at 9:19 AM, Bill Michaelson wrote: I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. In the overall scheme though this is really a minimal cost compared to dealing with issues that may arise over having a fully integrated network. We also only install managed switches and do have seperate vlans. The vlans may be either port based or tagged. In the last five years of doing VOIP installs, we have only had one customer the refused to add the second cable, and they were also the most unhappy. They also demanded the lowest cost phone option (IP301) and a Snom for an operator console. It all worked, just not very well, and ultimately they relaced it all. I n the real world, there usually are very inexperienced people using and managing the network. What is trivial in the data side becomes critical on the voice side and since most networks are run by the data guys, having it as seperate as possible really helps keep it all working well. One of the not so obvious issues is when the data guys are having a problem and go around rebooting things, dropping phone calls. On this list we tend to only think about the voice side, just keep in mind any data operations which are also going on. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Current Open Source Billing Package
After spending a couple hours scanning for an open source (non- commercial) billing package yesterday I am underwhelmed. Almost all of the packages listed on the WIKI appear to be defunct, for several years now. I will be happy to get a login and edit them out if that is the proper method to do so. My requirements are very minimal and at this point unless I have missed something will just write my own. I do not do calling cards. I have no near term need for the package to actually talk with asterisk at all, other than to import the CDR either via files or as a login to MySQL. I do have monthly recurring charges which need to be included monthly. I do occasionally have need to one off (manual) billing charges. Rating for calls would be nice but not mandatory ( we have very minimal International). Ability to export to an accounting package a plus. Ability to generate hard copy Invoices and/or email them to the cust. Ability to generate a list of current Invoices. Runs on Linux. All in all not a very complex set of requirements, but the few packages that seem to be currently offered generally do not fit the bill. Yes there are many commercial packages, but unless they are very minimal in cost I have no interest in them. So my question is, have a missed a golden nugget out there? tia Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Oct 29, 2008, at 12:30 PM, David Gibbons wrote: Fair enough, I guess I was concentrating on this line in Jerry's message :) The only reason I can think of not to is to eliminate the cost of the second cable. I believe you're mistaken about the QOS though. QoS is not required on lightly loaded links and will do nothing for you on over loaded ones. QOS will absolutely allow voice traffic to pass with priority over heavily loaded links -- this is in fact the reason that it would be implemented. Obviously giving priority to the voice traffic on these heavily loaded links serves to mitigate both latency and jitter. The concern is almost never one of taking bandwidth away from the desktop, but one of the desktop taking bandwidth (especially by introducing latency) away from the phone. Agreed -- but with VLAN tagging and QOS, the issue of how much bandwidth the desktop uses and/or needs becomes moot since the phone is given priority. Dave David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. --snip-- In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. --snip-- That's two _logically_ separate networks. The key point is that the last yard cable to the phone is not shared with the computer. The issue is not a lack of bandwidth but that the phone has to try and get its little packets inserted between the massive packets of a database lookup or file transfer in a timely manner (latency and jitter). You might get away with a single logical network on a smaller site or a larger one with very light traffic. QoS is not required on lightly loaded links and will do nothing for you on over loaded ones. I only use it on WAN links where bandwidth is more expensive. Allow me to clarify. Yes I do advocate seperate cable runs for phones and computers. Do not care if they both use a single switch as long as they are VLANd on seperate paths, either port based or tag based. And before everyone starts up again - :) - let me say that YES, I do install single cable fully integrated systems - when I manage the network. If I remember the OP was looking for real world examples and guidance. In the real world, just last week I picked up a new customer, drove 6 hours to a branch office of theirs that kept complaining about voice performance, and threw out the hub I found they had installed when they moved into their brand new building. Had a nice new switch - which I was told about - for their pc's. But all phones were on a hub - which I had not been told about. The new switch had been sent down to plug the phones into, but yeah. So in the real world I really like the KISS principle. Of course if there are qualified data folk ALWAYS makeing sure network is setup properly then feel free to disregard. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
I can think of two valid reasons to physically segregate the networks: 1) Insurance. I.e., to eliminate the possibility that otherwise properly configured QoS mechanisms become broken, either by accident, incompetence, or badly-designed or rogue software or hardware - or are otherwise handled carelessly as Jerry Jones suggested. But this is not a compelling argument to me in any but the most critical scenarios such as public-safety applications, etc. or you wish to eliminate service runs - that is unless they are always billable and your customers do not mind you informing them they messed up again and that is why they ahd issues. This is ok once or twice but some customers just cant control things and IF possible to reduce areas where problems could arise why not. 2) Customer preference. If you need the business, then the customer is always right. You might not have adequate credibility with the customer or influence over the design decision, and if a customer in such a situation gets it in their heads that voice and data can't coexist on wires, then it can't. True - just refer to my earlier examples. it is definately smarter at times to walk away. There is a variety of opinions, but no general consensus about where QoS failures typically occur, when they occur. I'm wondering if anyone has anyone has ever experienced QoS issues caused by contemporary Polycom phones like IP330s that had workstations hanging off their builtin switches? If you did, were you able to identify the cause, and was it due to any inherent failure of the phone, such as not marking packets or prioritizing dispatch correctly? No. Well other than the port going dead or flaky. But the switch had best be up to the task. I find in installs where customer is looking for inexpensive phones, they tend to want very inexpensive - and normally unmanaged switches. I will not install an unmanaged switch for other than a residential install. Plus even in fairly large installs where they are hitting an ITSP and traversing say a Watchguard firewall, the firewall will honor marked packets but cannot itself run diffserv and apply a tag. In this case the users pc's are in total control and all that corporate data and voip gets to compete with users streaming music et al to their desktops. In this case unless there is a local voip server even their inside calls will suffer. But the proper solution is to always have a firewall/router than can properly dispatch the packets to the WAN. Have a couple Juniper firewalls I hope to try in a couple weeks to see how they perform. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?
On Oct 27, 2008, at 3:01 PM, Andrew Kohlsmith (lists) wrote: On October 27, 2008 02:01:43 pm Jeff LaCoursiere wrote: Speaking of fring, I just got my brand new iphone 3G. Anyone have any comments on how well fring or any other sip client (siphon?) works on iphone? I do not like fring. It's buggy, it's unstable, it looks goofy -- but I have to say that yes, the SIP client appears to work. It won't reinvite off of their servers, though, so your audio path goes through them all the time. I need to learn how to write iphone apps and just write a simple straightforward SIP phone for it. -A. I second this. Yes it is difficult to find a plain SIP client not tied to a service. siphon is the best I have found so far. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is there a reference guide to pri debug span messages?
On Oct 23, 2008, at 3:10 PM, John Cheng wrote: Maybe I just haven't thought of the right google search terms -- but is there a website/guide out there that will help me understand the output from pri debug span? ___ perhaps this might be helpful? Q.931 Spec___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Latency woes, qos the fix?
On Oct 19, 2008, at 1:21 AM, Alex Balashov wrote: Stephen Reese wrote: Does the latency remain more or less the same regardless of the bandwidth load on the pipe? If so, TOS bits (what you refer to as QoS) won't help you. You've either got network issues (very likely if you have an intra- network ping of 30 ms) or the outside host you're sending the traffic to is just that far away in latency terms. Interesting. Just to clarify, the computer I'm pinging from is on the same switch as the phone so I don't see how there could be so much variance since the remote Asterisk server is on a very fast pipe. I've seen several people post that they have well under 100ms response. Is the time that the Asterisk displays just a ping to the device or are there more calculations? Any ideas besides TOS since that will not help much as you mentioned? Theoretically, the time Asterisk displays is just the result of round-trip time for a SIP OPTIONS ping which results in some sort of SIP feedback. In practise, that often ends up being considerably longer than the ICMP ping time and is often a very specious metric that does not give any real insight into the true end-to-end latency for media relay etc. Some of that has to do with the speed with which the far end's UAC core responds, so application-level latency as well as other latency within the propogation of the request up the stack plays into it. It may also have to do with inaccurate and/or wandering timing resolution used within Asterisk to time the return of those requests, especially if it depends on any kind of heavily locked threaded processes or other unknown event intervals. I do not know the answer to that. What I do know is that the time Asterisk shows for its 'qualification' pings can be 100+ ms higher than the ICMP round-trip time. I would use ICMP echo (traditional pings) to figure out if the latency is really the problem. The TOS field is meant to solve contention issues on the upstream path because routers that are set to differentiate between various DiffServ code points can packets marked as being of a certain class at a much lower contention ratio, depending on what else is enqueued. In practise, that means media can receive higher packet dequeueing priority if it contends with many other types of packets for upstream bandwidth. It won't help you on the downstream unless your provider is doing DiffServ tagging and your edge router is set to recognise the right bits and forward the packet on. But unless you've got that kind of setup going, you can't affect the contention of the traffic that is transmitted to you from somewhere else. As far as figuring out the true essence of the problem, ICMP pings can probably help to diagnose it along with accurate bandwidth usage measurements on your upstream pipe. Of course, the problem could also be caused by interface errors, duplex mismatches, bad cables, bad NICs, bad WICs, and just about anything else that can cause network problems that may not be easily detectable with conventional data applications but show up in real-time ones such as VoIP media relay. Alex is correct. Always check thereare no half-duplex links in your path. If you have an older dsl/cable modem or router that only has a 10M ethernet, it is probably half. Also make certain there are no hubs in the path. Keep in mind that colissions ar NORMAl for a hlaf duplex connection. TCP traffic simply retransmits, but voice (on asterisk) is RTP/UDP and the packet gets dropped. Even if it were TCP there is no time for a retransmit to be detected and resent. Using ehternet to detect the collision it does get resent, but there comes your jitter - which has much worse effects than simply latency. As far as measuring latency, doing a sip show peer andlooking at the qualify times is a GUIDELINE. It is my no means a correct indication, the real time can be much lower. I have noticed various ATA on the same networks as Polycom phones wil have sub 20ms times and the Polycoms will be 50ms. Yet all is as it should be and working great. Generally QOS will help with packet loss and jitter. Hope this helps. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones lose contact
On Oct 17, 2008, at 5:14 PM, Paul Douglas Franklin wrote: When off site, our IP phones lose contact after a few minutes of inactivity. They no longer receive calls, though they can call out. Asterisk acts as if it is ringing the phone, but the phone does not ring. The phones are behind a NAT/firewall. What is the most reasonable solution? qualify=yes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
The times they are a changing - or something like that. while gb on phones is not the norm today, it s becoming more so on the higher end flavors and will continue to do so since the life span of your switches will be several years, thinking ahead is a good thing my only concern is having too many poe ports in a single switch, especially if it is a 1U model, running many with 24 ports poe I have had failures after a year or so. And with the new POE+ spec coming this will get even worse. Think adding more fans = more noise to get rid of the additional heat they generate On Oct 6, 2008, at 12:04 PM, David Gibbons wrote: Right, it takes some doing to find a 1Gb switching phone though we ended up going with a system based on the Cisco 7941G-GE. This model supports all of the needed features including vlan tagging and 1Gb switching. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Robert Augustyn Sent: Monday, October 06, 2008 12:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] PoE switch recommendations? Most phones support only 100M switching though Unless you run separate cabling for VoIP and data but then you would not need the 1G uplink. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gibbons Sent: Monday, October 06, 2008 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PoE switch recommendations? Obviously we don't need 1Gb connections for VOIP :) Phones support pass through to the desktop and VLAN tagging. The need for 1Gb ports comes from wanting to have 1Gb at the desktop. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Monday, October 06, 2008 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PoE switch recommendations? On Mon, 6 Oct 2008, Ken D'Ambrosio wrote: Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't care much if it's in a single chassis or not, so long as it has Gbit uplinks. I'm curious as to why you want Gb uplinks on the switches? If we assume 100Kb/sec per phone .. (gross rounding, using 100Kb/sec per phone, rather than ~80 - make the sums easier and builds in a margin) 10 calls per Mb/sec. So for a 24-port switch, 24 phones all talking to 24 extensions off that switch, the max the uplink port is going to be pushing out is 2.4Mb/sec. For 200 extensions, say 9 x 24 port switches, with a single top-level (non PoE switch) switch with the PBX plugged in along side the 9 downlinks, that single PBX link will be carrying 2.4*9 = 22Mb/sec if all phones are in-use at the same time (and the PBX is carrying media) Now you may not want to build the network like that, but it seems that Gb is overkill just for the VoIP side of things. (And with that many extensions, I would suggest keeping all the phones on one set of switches) (Then again, it might not be possible to get big PoE switches without Gb uplinks, so it might be a moot point!) So satisfy my curiosity - why Gb uplinks? Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
On Oct 6, 2008, at 1:53 PM, Steve Anness wrote: I know I have asked about this before, but I thought that I would ask again with some more detail and maybe someone will have an idea. This is my first time to be setting up an asterisk server and I have a server running. I sent Linksys PAP2T’s to several remote users. Only one out of the four users actually work like they should. One of the other users I am assuming is behind a firewall on his wireless router and needs to open up the proper ports. However, I have two users in New York on a DSL connection and I can’t understand why things are happening like they are. Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be “unreachable” and I can no longer call; however, they can still make calls. do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
Google works enter this along with your search string site:lists.digium.com your.search.string.here dont type the On Sep 29, 2008, at 2:42 PM, Brian Webster wrote: What is the best-recommended resource for searching archives of this mailing list? Thanks for your time ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
On Sep 29, 2008, at 9:55 AM, Yehavi Bourvine wrote: Try AudioCodes MP-124 which is 24 ports FXS. I have one but haven't used it much yet, so I cannot comment about its quiality. \ Sorry, cant agree with this, tried a couple and replaced with channel banks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selectively disable echo cancellation?
When the cards hears the fax tone it should auto disable the ec.
On Sep 2, 2008, at 9:42 PM, Octavio Ruiz wrote:
On Tue, Sep 2, 2008 at 6:16 PM, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
Hi, all. I have a Sangoma A104D (on-board, DSP-based echo can); I'm
currently passing through some of my in-bound calls to a legacy PBX
(which
I hope to eventually replace). That being said, until I do, I'd
like to
kill echo cancellation for the passed-through calls -- I don't want
to
mess with their fax reception.
Any idea how to do this?
Is
echocancelwhenbridged=no
inside zapata.conf what are you looking for?
If not, what I figured out is if you run
System(wan_ec_client wanpipe1 disable ${VALUE}) ;
in your dialplan logic [perhaps inside a macro called with the M()
option for Dial()] would do the trick.
Don't forget that you obtain Zap/${VALUE}-1 from ${CHANNEL} (using
some variable stripping) and to run
System(wan_ec_client wanpipe1 enable ${VALUE}) ;
at Hangup.
Regards,
--
Octavio H. Ruiz Cervera
Tel.: (+52 55) 8590-9000 Ext. 7016
Mobile: (+52 1 55) 14-087790
Mobile: (+52 1 55) 41-351242
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Re: [asterisk-users] Remote Support
On Jul 28, 2008, at 5:50 PM, Jason Parker wrote: Philipp Kempgen wrote: I would suggest screen ( http://en.wikipedia.org/wiki/GNU_Screen ). screen doesn't solve the security aspect of your question though. Grüße, Philipp Kempgen Actually, it could. What I've done before, is give out an unprivileged account on the box (or some intermediate gateway box). Once they log in, you ask them to run screen (as the unprivileged user) to connect to a session you've created, then proceed to login as root yourself. If they disconnect their screen session, they leave your root terminal. You can also kill the screen session at any time. _ If you have X running you could also do VNC which would let you see what they are doing. Perhaps just change run level when they need access? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two way bandwidth test
On Jul 16, 2008, at 3:11 AM, Femi wrote: If you can get a machine at the other end of the link you could use the Mikrotik bandwidth tester You can find it here - http://www.mikrotik.com/download.html Femi or just run iperf on each end http://sourceforge.net/projects/iperf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Telco MWI with Asterisk 1.6-beta9
On Jun 22, 2008, at 6:51 PM, Kevin P. Fleming wrote: Jim Duda wrote: My Telco service is Verizon FIOS. I know that MWI is working, because if I pick up an analog phone set attached to the line, I can hear the stutter tone. The MWI detection in chan_zap/chan_dahdi is not for stutter tone; it supports either FSK MWI pulses (slimier to Caller ID, but transmitted during idle times on the line), or neon-style pulses of AC. Unless your telco service specifically supports phones that can display whether you have new messages (and how many), you don't have the right kind of MWI signaling from the telco. It is doubtful that *any* residential phone service will work in the way that chan_zap is expecting; FSK and neon MWI signaling are generated by legacy PBX systems with analog ports, not telco (CO) switches. Actually CLASS MWI, ie FSK, is a standard feature of Telco COs. It may be used with stutter, they are not exclusive, check with Verizon. I designed and manufactured these 20 years ago and Kevin is correct it uses same technology as CallerID known as CLASS. The analog styles, which I also designed and manufactured, are only found on PBX systems. There are several flavors that all signal based on some voltage mechanism. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SIP and DHCP problem
On Jun 9, 2008, at 2:29 PM, Lyndon Griffin wrote: Apologies - I know this isn't either Polycom or ISC support, but if anyone would have an answer to my problem, I'm certain they would be on this list. I'm experiencing odd behavior with Polycom handsets obtaining DHCP addresses. It always worked fine for me up until a few months ago. Unfortunately, I can't narrow down when it stopped working, or why. All my Polycoms now appear to ignore my DHCP server entirely, according to the following pattern: Polycom - DHCPDISCOVER Server - DHCPOFFER on the correct network Polycom - DHCPREQUEST on the wrong network Server - DHCPNAK Polycom - Rinse, repeat ad infinitum Had the same issue a year or so ago - it related to a code version on the Polycoms. We wiped the flash and let them reload software I think. dont think we changed code but that took care of the issue. This was on one of our IP430 installs, never had it happen with 6xx series - yet. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE budget
On Jun 7, 2008, at 9:51 AM, Rob Hillis wrote: On the Linksys side, we have a load of SRW-224P switches out in the wild powering 24 Snom 370s (around 7W each) off each switch. Likewise, we sell these things by the bucket load and have no problems powering phones from all 24 ports. Just curious - have these ever gotten quieter? We installed one when they first came out and it was WAY to loud for an office environment, data center would be OK. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE budget
On Jun 5, 2008, at 5:08 PM, Bill Michaelson wrote: I'm considering using a PoE switch like this... http://www.tigerdirect.com/applications/SearchTools/item- details.asp?EdpNo=3023334CatId=2800 ...to power as many as 24 Polycom phones of varied kinds. The sales lit indicates 190 watts available for PoE devices. But I'm concerned about a problem someone reported elsewhere... They said... -- -- Is there a reason that Polycom phones do not support PoE classes? We ran into a scenario recently where we could only power 11 Polycom 550's on a 24 port switch. This is because the Polycoms do not announce themselves as being in a specific PoE class, even though the phones only need 6W the switch assumes they need as much power as possible and allocates 14.5W to each port. We have had to resort to running unsupported firmware on the switch to get it to power 24 phones. -- -- Does anybody here have insight about this? have used many fsm7326p to power 24 phones or 726tp to power 12 phones and they work great ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
And you are using g.711 so the sounds are passing correctly and not
being distorted? Try calling a person and pressing digits to verify
they are inband during call?
On May 5, 2008, at 4:31 PM, Jason Wolfe wrote:
Yes, and I verified watching the output that it was reading the
new .conf
file.
jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-
[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Monday, May 05, 2008 4:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF
Did you remember to do a reload in the Asterisk CLI?
Jason Wolfe wrote:
Ok, I removed the T/t/w/W options but unfortunately it is
still
responding
the same way.
Ps. I have no options set on the dial() function now.
jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-
[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Monday, May 05, 2008 4:31 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] DTMF
Remove the T/t/w/W option from the Dial line.
Jason Wolfe wrote:
Ok, ever had one of those issues that you're
sure is
quite
simple to solve
but you can't seem to get anything useful from
Google
or
anywhere else and
so you're ready to throw your computer out the
window? Well,
I'm there!
I am using a simple Zyxel VoIP phone to dial
outbound
calls to
a PSTN
termination provider, so my extension file is one
command.
Dial()
Anywhere I call I probably need to enter an
extension, but as
it should,
asterisk tries to respond to these key presses.
How
do I pass
the DTMF tones
through so that I can navigate the IVR of the
system
I'm
calling???
--
Consulting for Asterisk, Polycom, Sangoma, Digium,
Cisco, LAN,
WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design.
Based near
Birmingham, AL. Now accepting clients worldwide.
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Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN,
WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide.
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Re: [asterisk-users] Need some input for Quad T1 and channel banks.
On Apr 2, 2008, at 9:22 PM, Al lists wrote: Bad memories from AudioCodec :) Second this. My favorite is Vega, but they have terrible support in US. Have many Adit600 connected via Digium T1 - work great. Even FAX if PSTN PRI connected to same card. And no the Adit600 is not a switch, hence it does not support PRI signalling, it will pass it through and perform DACS functions very well. You can also buy their CMG card and turn into a gateway, but it will be MGCP not SIP. If using as a channel bank I strongly recommend their newer FXS-C cards, they support line testing and are the only diagnostics available for the FXS ports. Also I hear Carrier Access is now Turin Networks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing off-hook with Polycom SoundPoint IP 430
What does your digitmap on your phone look like? This is what
controls sending the call to * when it recognizes a complete dial
pattern. The phone does not send digit by digit. If it is waiting for
you to press send, then it does not recognize your pattern.
On Mar 26, 2008, at 8:18 AM, Brig C. McCoy wrote:
Hi…
I’ve been fighting this for a while now, trying clean builds of
Asterisk 1.14.18, 1.14.19rc3, and then 1.6 Beta 6 today.
No workee. L
Here’s the results for various calls made off-hook (push the blue
Speakerphone button on the Polycom 430):
988852700 – Phone waits for me to either hit the soft-key “Send” or
“EndCall”. If I hit “Send”, it dials through with no problem.
98168852700 – Before I get the last “0” pressed, the phone presents
me with a second dial tone and a prompt at the top of the screen,
“Enter more digits”. Asterisk console presents
“== Using SIP RTP CoS mark 5”
917852963296 – Before I get the “96” pressed, results as
immediately above.
If I dial these numbers with the phone on-hook, and press “dial”
they work fine.
If I modify my dialplan to remove the dial nine requirement, all
three methods of dialing out, off-hook, work fine…although I do
have to press “Send” when dialing 8852700.
The seemingly relevant portion of the dialplan is as follows:
;
; BEGIN - Outbound Call Handling
;
;
[outbound-local]
exten = _9NXX,1,Dial(${TRUNK0}/${EXTEN:1})
exten = _9NXX,n,Congestion()
exten = _9NXX,n,Hangup()
exten = _9NXXNXX,1,Dial(${TRUNK0}/${EXTEN:1})
exten = _9NXXNXX,n,Congestion()
exten = _9NXXNXX,n,Hangup()
exten = 911,1,Dial(${TRUNK0}/911)
exten = 9911,1,Dial(${TRUNK0}/911)
[outbound-long-distance]
exten = _91NXXNXX,1,Dial(${TRUNK0}/${EXTEN:1})
exten = _91NXXNXX,n,Congestion()
exten = _91NXXNXX,n,Hangup()
[hang-up]
; Hang up
;
exten = s,1,Playback(thank-you-for-calling)
exten = s,n,Playback(goodbye)
exten = s,n,Hangup()
;
;
;
; END - Outbound Call Handling
;***
The only difference between the Asterisk versions is the presence
on the Asterisk console of an error message with Asterisk 1.4.18
and 1.4.19rc3, which is similar to the one noted on the forums:
“NOTICE[6145]: chan_sip.c:13795 handle_request_invite: Failed to
authenticate user 6000 sip:[EMAIL PROTECTED];tag=whatever it
was” I do not see that error message on the Asterisk console for
1.6 Beta 6.
The forums note which seems in the neighborhood is at
http://forums.digium.com/viewtopic.php?
p=63872sid=aff61bbd5ddeea61bc831239b220db23
Anyone have any bright ideas on what might be wrong and/or
troubleshooting tips?
…brig
--
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816-767-5549. This will help with issues getting full exposure to
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Brig C. McCoy
IT Help Desk
ThyssenKrupp Access Corporation
4001 East 138th Street
Grandview, MO 64030 USA
Phone: +1 816-767-5577
Fax: +1 816-765-6459
Email: [EMAIL PROTECTED]
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Re: [asterisk-users] Newbie Polycom: IP601 console with expansion module
The Polycom will display a different icon if DND On Mar 12, 2008, at 2:04 AM, Lee, John (Sydney) wrote: Special dialplans for reception are entirely up to you. The only reason reception phones have different dialplans to normal extensions is that often people want the receptionist's phone to behave a little differently. Thanks Rob. I talked to the receptionist this afternoon. She said it would be great if the expansion module could show whether a staff is engaged on the phone or whether the staff has turned on DO NOT DISTURB (BTW, can Asterisk flag a phone as DO NOT DISTURB?). Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Background Noise Elimination
On Jan 7, 2008, at 6:19 PM, Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Norman Franke wrote: Greetings! We have a somewhat noisy background in our call center, and I'd like to reduce this. Obviously, we could plaster the walls with sound absorbing material, but is there anything we can do in software either using any algorithms for our open source-based SIP library or inside Asterisk itself? Related to this, anyone have a good source for good panels? We are using Plantronics noise canceling headsets, which don't really seem to work all that well. Our ancient system handled noise much better, but I suspect that was partly due to the Dialogic ADPCM algorithm used that just reduced the intelligibility of lower volume noises in general. We are using PCMU direct from the agent's mic to through Asterisk to PRIs, so we don't suffer from compression artifacts. The down side, is that you can make out even very quiet conversations in the background (like 3 agents to one side.) How have people handled this? I'm experimenting with a noise gate that will lower the volume when the agent isn't talking, but that won't help when the agent is talking. Nah, there's nothing really. The noise gate is your best bet. I would assume that while an agent is talking the customer will be listening to the agent, so the background noise will hardly be noticeable. The issue is, while two people are talking its pretty hard to remove just one of them from a wave file. Try the noise gate and see how you go. Oh, you might want to try a downwards expander instead (a noise gate but with ratio as well as threshold). We have an IP600 located in our colo, a very noisy environment. For a spooky experience make a phone call and pass the call through a Ditech audio processor in the path of the PRIs. You will hear no background noise. You can even use the speakerphone. Even Polycom to Polycom is not too bad. But an all IP path to anything else and you cant hardly hear the other person. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Digit Map
On Dec 31, 2007, at 11:36 AM, Michael Munger wrote: I need the digit map to call China. Example number: 011-86-10-6887- 011-International (obvious) 86 is country code (China) 10 is city code (Beijing) Last 8 digits are the number. I tried using 011xxx.T but it always asks me to enter more digits. Tried some variations as well, but no joy. Yours should work if you wait long enough for t to timeout. How about 01186xx? Plus, IARC, when dialing offhook, pressing # should terminate dialing and send what it has at that point. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One server, multiple companies
[incoming]
exten = 2125551211,1,GoTo(companyA,1)
exten = 2125551212,1,GoTo(companyB,1)
exten = 2125551213,1,GoTo(companyC,1)
[companyA]
exten = 2000,1,Dial()
[companyB]
exten = 2000,1,Dial()
[companyC]
exten = 2000,1,Dial()
On Dec 13, 2007, at 5:53 PM, Diego Andrés Asenjo González wrote:
-- Mensaje reenviado --
From: Eric C. [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Sun, 9 Dec 2007 19:55:51 -0500
Subject: [asterisk-users] One server, multiple companies
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three
distinct phone systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan
within the extensions.conf file. I'm using
exten = _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10})
to determine which number is being dialed by the caller and then
using a gotoif to get to correct greeting (correct company).
My question is... lets assume all three companies have extension
numbers being 2000, 2001 2002, how does one separate them?
Or, lets say the extensions are:
company A -- 2000, 2001,2002
company B -- 3000, 3001, 3002
company C -- 4000, 4001, 4002
Since they're on one server with one asterisk process, how can I
use context correctly so that the user at 4002 cannot get through
to the user at company A whose extension is 2000 as currently, I
can dial 2000 from phone 4002.
That's my current problem, how should this be setup? Is my
architecture correct? Should I be running different processes for
each company? Can context resolve what I need?
Hi,
You should try DeStar, a management interface for Asterisk:
http://destar.berlios.de/
DeStar supports Virtual PBXs, then you can install it and take a
look at the dialplan. Sorry for the late answer but I've just read
the list messages.
Bye,
Diego Andrés.
So
Please advise.
thanks,
Otto
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Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
On Dec 10, 2007, at 7:45 AM, Michael Melia Jr. wrote: I haven't found outcall that confusing though I do agree that a TAPI Driver that makes use of the available outlook call functions will make for the easiest, most streamlined user experience. I also agree that these convenience and little feature are very important especially with Microsoft entering the VOIP platform market with a product that is sure to integrate with the users desktop and office extremely well and in every aspect. Asterisk need this functionality to stay competitive on the end user experience front. I do like the concept of using sip-tapi. I looked at a couple years ago and if I remember coorectly it communicated via SIP vs the manager interface. Much better in my opinion. Although I never was able to get it working correctly, then again I dont do windows. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
I will miss them. It was nice having a local company with a few Polycoms in stock most of the time. A month or so ago we needed some quick and were unable to contact them, either through their toll free or local numbers. I swung by their office last week and nocticed it was vacant. On Sep 27, 2007, at 1:49 PM, Darrick Hartman (lists) wrote: Doug wrote: http://www.atacomm.com/ ATACOMM Dear Atacomm Customers, We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm and its parent company Ataractic Corporation has ceased operations. We appreciate the 7 years of loyalty and support from our customers. We sincerely regret any adverse effects this may have caused. I'd say that's pretty self-explanatory. My credit card company is trying to recover about $800 in fraudulent charges for duplicate transactions and failing to send the merchandise for a transaction that dates back to late August. Normally I'd say this sort of thing belongs only on the biz list, but this sort of issue may affect so many people it's worth noting here (but not dragging out with hundreds of me toos). -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux-HA and Asterisk
How about 20+ on a Qwest DSL modem hitting our server? Works great. On Sep 12, 2007, at 7:23 AM, Dovid B wrote: Eric, Try 5 polycoms behind the same NAT router. Let me know when you grab a drink ;) - Original Message - From: Eric ManxPower Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 12, 2007 2:43 PM Subject: Re: [asterisk-users] Linux-HA and Asterisk Polycoms work just fine behind NAT. Mike Clark wrote: Chris Mason (Lists) wrote: Mike Clark wrote: Yes, the Asterisk boxes were on private addresses. The Polycoms are also behind a NAT. Yes, I tried using externip in sip.conf and this allowed registration, and calls to be placed, but no audio. Unfortunately, Polycom does not support STUN. Your problem is not Linux-HA, it looks like that is fully functional. Your problem is the same one many people come across. You can't put Polycom phones behind NAT, it won't work. If you have to have the phones behind NAT, which I advise against, use Linksys which probably work better, and use a SIP aware NAT device. Better still, put the phones on the same network as the Asterisk PBX and say goodbye to your problems. Thanks Chris. Unfortunately, these solutions aren't an option. I guess I was hoping someone had found the silver bullet or some undocumented Asterisk feature that solved the issue. Back to the drawing board. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
On Sep 11, 2007, at 7:29 AM, Eric ManxPower Wieling wrote: Rizwan Hisham wrote: well he does not have access to hi sip settings, so he cant edit the host=differentIP every time he moves or registers from anyother place. Actually he should be able to register from anywhere in the world but once he has registered with us, i dont want anyone else to register with my asterisk using his credentials. Then make sure nobody else knows his credentials. This isn't rocket science. How exactly do you propose to determine of the user moved the device to a new location .vs. a 2nd device trying to register with the same credentials. In any case, Asterisk does not have any facilities to do what you want to do. How about some method of checking for a current registration when a new one is received. If presently registered at a different IP then disallow new attempt. No I dont know any existing tool within * to accomplish. Anyone else? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low-level dump for PRI dchan debugging
On Aug 9, 2007, at 9:37 AM, Erik Anderson wrote: On 8/6/07, Erik Anderson [EMAIL PROTECTED] wrote: I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending reset all channels signals to my system, to which he's getting an establish remote response from my asterisk box. I've been running a packet dump (wanpipemon -i w1g1 -c trd) of my d- channel this whole time and have yet to see a single incoming packet. I believe I *should* be seeing an incoming packet when he sends the reset, correct? Is there any way to do a completely raw dump of the d-channel? Thanks to everyone who offered suggestions on how to troubleshoot this issue. After working with the telco for over a week on this, I finally got them to admit today that they have a configuration problem. I had been telling this since day 1, but they didn't listen to me. Their change in perspective came when they had a tech come on-site with a PRI emulator device. He connected that directly to my asterisk server and was able to make calls with no issues whatsoever. Fortunately after this final test, they admitted that the problem must be on their end. Hopefully they'll get it sorted today. As an aside, I had a quick question regarding smartjacks. Is there a jumper or something on the smartjack itself to change from an old-style EM T1 to a PRI? I'd think that change would happen in the telco's switch, but I just thought it might be a possibility. In my case, as I stated in my original email, the bchannels come up fine, but not the dchannel. This makes me think it could be something simple... It will be something simple, like getting a clueful tech on their end. No the smart jack has no bearing on d channel. Old style or new style the T1 is used however the gear on either end says it should be. The smart jack just passes info through itself. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Learn some terminalogy before mounting this task.
On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote: All, It's been some time since this thread was alive but we are now seeing some progress in this project. Which I will document. We have ordered a T1 for the new building which we are moving (We are getting 14 channels of the T1.) and have a Sangoma A101 card for a 3U rack server. The T1 will have B8ZF decoding and ESF framing which the sangoma card should handle. They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ? Yes. You want NI2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Marceau Sent: Tuesday, April 10, 2007 11:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Learn some terminalogy before mounting this task. James, I'm sorry that I can't add anything but just wanted you to know that I am watching this thread with great interest and suspect that many others will too. Thanks in advance for posting lots of details as you go thru the process. Pierre [EMAIL PROTECTED] 4/10/2007 10:41:36 PM Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS handsets) ports, etc. Would I be better off getting a Zapata T1 card and forgetting the Channel bank all together(Use the digital signal)? You could do that. Personally, the easiest approach I would say would be to order a PRI. They've probably considerably gone down in prices, too, especially if you go shopping with some friendly CLECs. The rule of thumb in the industry is that generally, once you pass the threshold of six or seven POTS lines, it becomes economical to just order an entire PRI, and once you do that, there usually aren't *very* considerable savings to be gained from turning down all but a few channels. A PRI has 23 channels (bearer channels (B channels)) and one signaling channel (D channel); it's a type of T1-based ISDN interface. So, you might potentially be able to get 23 in/outbound phone lines for roughly the same cost or a modest increase, which would increase your organisation's capacity to do things like conference calling and other things which tie up large amounts of outside lines. Do beware that if you go this route, PRIs can be ordered as inward-only (typically used for modem and termination-only telephony applications like voicemail, IVR, conferencing, etc.) or bidirectionally. If we go with a Zapata T1 card for the Asterisk server would we be able to provision an analog phone line, for say a FAX machine from it? No, not if the card doesn't have FXS ports on it. But you could get another Digium or Digiumlike card that does, even if it's just a single-port (like the hugely popular
Re: [asterisk-users] FXS channel bank
On Jun 28, 2007, at 8:00 AM, pixiesfr wrote: hello, We looking for a channel bank to connect 120 analogs phones, in SIP to an Asterisk .. Did someone have some product in mind. A channel bank must connect via a T1 by definition, which would then give you 24 phone lines per T1. This would require 5 T1 connected to your asterisk server. OK 4 if E1 as it probably is in your case. However with your requirement for SIP you are looking for a gateway to connect your phones. Most are 24 port, though some are 48 port. Names to look at would be Carrier Access, Audiocodes, Vega etc. I do like the Vega unit except for their support - or lack thereof - here in the US. They do have both 24 and 48 port units. Your other option would be to do GR303 which would allow you to hang many lines off a few T1/E1 circuits, except it is definately not SIP. If phones are not at location of your asterisk server and you really want to do sip, it may be simpler for this many phones to install an additional asterisk server at the remote location and install a quad port T1/E1 card and hang channel banks off it. Good Luck ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inexpensive Layer 3 Switch?
You do not need an L3 switch for this, just any managed switch which does vlans Unless there is something else? On Jun 26, 2007, at 12:07 AM, Marty Mastera wrote: Any recommendations on an economical layer 3 switch? Preferably something that you have hands on experience with connecting to IP phones with attached PCs? Specifically I need the ability to set the VLAN in the phone to tag voice packets and to set a native VLAN on a per port basis on the switch to put the untagged packets from the attached PC into a separate VLAN. POE is not a requirement but if you have suggestions for an economical layer 3 switch with POE I’d be glad to hear them…so far I’m looking at the SFE2000 from Linksys. thanks No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.7/868 - Release Date: 6/25/2007 12:20 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High Port Count ATA
You can add their gateway blade to convert to voip via ethernet, but it only does mgcp. How about doing GR303 to an access navigator with channel banks hanging off that? Pricey but carrier class gear and scales WAY up. Could also do Adtran total Access concentrator (4303?) feeding their total Access 1500 with TR08 would be more dense and possibly cost less. Best way is also going to be determined by how many calls up at one time. Going with one of the 48port sip gateways may be ok if locally peered with the Asterisk server. On May 31, 2007, at 5:49 PM, Douglas Garstang wrote: Cory, I’m not quite clear on that. Do these channels banks have an IP uplink port so that each FXS port can SIP register to asterisk? Doug. From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Thursday, May 31, 2007 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] High Port Count ATA Channel banks would work. Rhino works well, or if you need more chassis density, try the Carrier Access ADIT600 configured with FXS blades. Cory J Andrews From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Thursday, May 31, 2007 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] High Port Count ATA I’m trying to find a high port count ATA device. We have a lot (up to 110) analog devices that we need to bridge to IP. Rather than go out and buy 110 ATA’s, it would make more sense to buy a single chassis type device with some number of ports and blades. Anyone know if such a device exists? Thanks, Doug. ___ --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] Polycom Static IP
When turning of dhcp, dont forget to set all other attributes manually. Ones that would effect this are IP Address Subnet mask Gateway boot method tftp/ftp Server Address username/password if ftp vlan Assuming you are setting a hard IP for the server, if using a url then donot forget to add dns entry also On May 28, 2007, at 3:23 PM, Steve Totaro wrote: Sounds like a firmware bug, VLAN or other network configuration bug in the phone (subnet perhaps?) Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Forum Sent: Monday, May 28, 2007 3:56 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom Static IP I am still having issues with my Polycom 301 phones when I disable DHCP. I give the phone a static address and I keep getting the error ‘could not contact boot server using existing config’. As soon as I set it back to DHCP enabled the phone can see the boot server and I’m online. Steve ___ --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] Calls in ulaw, not gsm as desired
A simple glance at their website will tell you this about the 501 G.711 μ/A and G.729A (Annex B) configuration On May 2, 2007, at 12:22 PM, Jaswinder Singh wrote: Try ilbc if the phone supports (free) or g729 ( better but your asterisk will need license if you want to transcode calls from g729 to other codecs or want to record calls ) . Also check your phones config if its support multiple codecs . . On 02/05/07, Rob Schall [EMAIL PROTECTED] wrote: So I reloaded things and had just gsm set for 2 of my polycom 501 phones. However, the logs say No codec found, which leads me to believe that polycom 501 phones can't use gsm. Does anyone have something like this working? If not gsm, is there a better option with these phones over a low bandwidth situation? ___ --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 SP 601 Reboot Issue- Help!
The only reboot issue I have with 1 sidecar is the side car deciding to randonly rebbot, not the phone itself Perhaps upgrading to 2.1 will help? On Apr 24, 2007, at 10:51 AM, J French wrote: I have a Polycom 601 with 3 expansion modules running 2.0.3. We have Buddywatch set up on around 42 users on the expansion modules. We are experiencing reboots on the 601. Today it happened twice after users paged through the phones. The page groups have about 23 phones each. There is a third page group comprising all 46 phones. I'm thinking it may be an issue with changing buddywatch state on so many buddies so quickly. Also, the cpu usage is pegged at 100% for around 3 minutes after it reboots, FWIW. Anyone else experiencing rebbots on the 601? Advice is really needed! 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
Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggishkeys - new info
On Apr 12, 2007, at 1:49 PM, Kevin P. Fleming wrote: Got off the phone with Polycom on this I have the same problem with my new 601 phone here (haven't seen the problem on the 650). I am using an IP650 with the latest firmware (and the corresponding sip.cfg file) and I have seen this behavior. It is most noticeable when on-hook dialing, where I will dial two or three digits and then press the fourth digit and nothing appears on the display for 1-2 seconds for that keypress. I am using new files with my 601/sidecar I have the issue and agree with Kevin, though I do mostly use it on hook. I have also noticed the end call or speakerphone button to be inoperative at times. It definately appears they have a bug and are not reading keypress in a timely fashion. I have also notice the sidecar has resumed its frequent rebooting again, had died down somewhat with the 2.0 code stream, but is back more often now with the 2.1. The phone is fine, but the sidecar will reboot randomly - whether idle or on a call. ___ --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] Loudspeaker
Hmm - just received an email from these guys last week. I know nothing about them. On Apr 15, 2007, at 9:23 PM, cb wrote: On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? Do you already have the loud speaker? If not, I know there are various vendors of extension phone bells that do nothing more than plug into an analog line and ring the nice loud bell when a ring signal is received. You could easily combine one of those with a cheap ATA with FXS port. -chris www.mythtech.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 ___ --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] FW: Polycom 501 issue with latest firmware : sluggish keys
I have actually noticed it on my personal 601 after upgrading past 1.6.7 to 2.0 and 2.1 Yes it is still doing this and is very annoying. Hopefully Polycom will fix by next release. On Apr 11, 2007, at 4:33 PM, Noah Miller wrote: Hi Mike - Somebody was helpful enough to give me the very latest release of Polycom's firmware (2.1.0). Unfortunately, I still get that issue. So I'm stuck asking again: Anybody ever got that? I've got quite a few Polycoms of various models running in a number of asterisk installs. Some of them are on 1.6.7, but most are on 2.0.3 or 2.1.0. I haven't seen this one at all. I would definitely call your reseller to have them bring it up with Polycom. If your reseller won't take the time, you may be able to find others that will - if you buy a phone from them ;-). www.voipsupply.com comes to mind, but I'm sure there are other vendors who will go to bat for you. - Noah ___ --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] Polycom 501 issue with latest firmware : sluggish keys
It has nothing to do with actually dialing. Even trying to press end call or the speakerphone button does not work at times. Have tried removing side cars etc, but definately seems to be a bug in the 2.x code stream. On Apr 11, 2007, at 5:37 PM, Eric ManxPower Wieling wrote: Jim King wrote: I've seen an issue like this from time to time on 601s, even with the latest firmware. Not just the softkeys, but also the dial keys. The phones seem to run slow sometimes, failing to respond to a key press right away but getting to it eventually. It usually clears up after a few seconds. Also, I've noticed that the 601s sometimes ignore key presses altogether, just as you describe. I have not yet found a solution for this problem... Try setting this in sip.cfg: dialplan.impossibleMatchHandling=1 I suspect it is either 0 or 2 now. ___ --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] Multi-Level Queue
The reasoning behind all of this is that I want to ring desk phones and then if they don't answer, I want to ring cell phones. If I ring the cell phones too long, someone's voicemail will pick up, which I don't want. So if I set it up where they have to ack it, I can ring the cell phones and if someone's vm picks up, it is no big deal. Also the cell will answer with VM if it is turned off, out of range, etc ___ --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: OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED
On Mar 6, 2007, at 1:55 AM, Tomislav Parcina wrote: Kristian Kielhofner wrote: Hey everyone, I came across a situation where I needed to use CDP to advertise a voice vlan to Polycom/Cisco (and other CDP capable phones) without a Cisco switch. Hi Kristian! Thank you for your work. I'm not able to test this right now, but I'll sourly need this sometimes. Hmmm - might be me but I am unable to find the beginning of this thread. It does sound interesting. ___ --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] SOLVED: Call forwarding and 1.2.x
We had an issue, and I know others had posted the same on the list. Scenario: Polycom phone user sets call forward to a toll free number(in our case) Call arrives for the phone, the phone notifys asterisk, asterisk dials new number. Telco drops call. But if you dial direct to the number it is a good working number. Solution Turns out our carriers DMS had a tuple on the PRI set incorrect. Seems they did not like the call forward information element sent in rn format. Setting the tuple correctly solved the issue. But it took the carrier a call into Nortel to have them figure it out. Switch techs had never seen that tuple used before. Still not sure what the rn format vs any other is yet. Anyone know? ___ --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] Transfer Caller ID
Not sure about others, but on Polycoms a blind transfer sends original callerid, screened sends operators callerid On Feb 19, 2007, at 8:55 AM, Rob Schall wrote: I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system (actually the operator). The operator (a real person) answers the call and presses transfer on her polycom 501 phone. I see an incoming call From: Operator. After I pick up her call, she presses transfer one final time to complete the transfer. However, now that the call has been completed, it still shows From: Operator. I need it to show From: 213-555-1234. I tried setting the o setting in Dial, but that didn't seem to fix anything. Rob ___ --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] Transfer Caller ID
Not an asterisk setting. It is how the endpoints perform the transfer. On Feb 19, 2007, at 9:21 AM, Rob Schall wrote: I probably have a screened transfer setup. Is that just a setting somewhere I can easily change? I'm trying to avoid making users press extra keys, like #1 or anything like that. Rob Jerry Jones wrote: Not sure about others, but on Polycoms a blind transfer sends original callerid, screened sends operators callerid On Feb 19, 2007, at 8:55 AM, Rob Schall wrote: I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system (actually the operator). The operator (a real person) answers the call and presses transfer on her polycom 501 phone. I see an incoming call From: Operator. After I pick up her call, she presses transfer one final time to complete the transfer. However, now that the call has been completed, it still shows From: Operator. I need it to show From: 213-555-1234. I tried setting the o setting in Dial, but that didn't seem to fix anything. Rob ___ --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 ___ --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] Toll-free dialing via PRI problem
This is a common issue with large inbound call center operations. They like to cheat. They actually start sending prompts to the caller without actually signalling their carrier that they have answered the line. Typically they do not answer until a phone is ringing or you are in a queue. I do believe this is illegal per the FCC. From asterisk, you do not hear anything other than ringing as it does not cut the audio path through until it receives the answer from the far end, hence the steady ringing. This allows the large centers to reduce their billable minutes by enough to warrent them to try it. On Jan 31, 2007, at 10:51 AM, McGhee, Stefano wrote: Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear ringing but the calls are never answered. All other calls, and most toll-free numbers are not affected. The numbers that are affected are all travel related companies (United Airlines, American Airlines, US Air, Starwood Hotels, etc.) we cannot connect to any of these numbers. Hey Tim, All I can offer you is the fact that I see the exact same thing on my setup that uses * and a TE411P. I've also seen it when calling Lenovo tech support and Sirius Satellite Radio. On the latter two, it bypasses the auto-attendant when I call and connects me straight to an operator/technician. When you call on regular PBX or cell phone, you are greeted by an auto-attendant, press 1, yada-yada. Let us know what you find out. Cheers, Stefano ___ --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] How would you compare feature set to a Metaswitch?
OK I need some help. Looking for comparisons for a large customer wishing to provide voip service over a region. We are up against Metaswitch who is claiming they can do anything Asterisk can do. I do not have too much information on Metaswitch so am looking for any information, preferably real world experience on how Asterisk and Metaswitch would compare side by side. Thanks in advance. Jerry ___ --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] Getting confused on signalling mode Vs framing and encoding: T1 CAS
On Jan 24, 2007, at 10:20 AM, Thinselin, Vincent wrote: Hello, I'm trying to make my asterisk box to act as a telco, in order to reproduce a US environment in europe. Our telco provider is giving us those settings: ESF B8ZF Inbound = EM Immediate Outbound sig =Wink Start Yield to Glare = Yes Those trunks are using CAS for signaling. I have tried many configs/combinations in zaptel.conf and zapata- channels.conf In zaptel.conf, when having something like span=5,0,0,cas,b8zs and in zapata-channels something like signalling=featb try em_w: E M Wink Start I end-up in the log file with something like chan_zap.c: Got hook complete in MF FGD, waiting for wink now on channel 125 If in zaptel.conf I put something like span=5,0,0,esf,b8zs My call is immediatly stopped: Jan 24 17:17:33 DEBUG[25076] chan_zap.c: Got event On hook(1) on channel 125 (index 0) Jan 24 17:17:33 DEBUG[25076] channel.c: Didn't get a frame from channel: Zap/125-1 Jan 24 17:17:33 DEBUG[25076] channel.c: Bridge stops bridging channels SIP/2707-0083be10 and Zap/125-1 What is the correct zaptel.conf for my case ? If I specify esf in zaptel.conf, where do I mention I do CAS for RBS ? you dont - signalling type sets it Thanks. V.Thinselin ___ --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] connecting a FXS-to-sip 4 port device to an avaya system
analog station ports = fxs analog line ports = fxo, assuming 2 wire loop start On Jan 18, 2007, at 8:26 PM, Erick Perez wrote: Thanks Jerry. Are the avaya station ports a special type ? On 1/18/07, Jerry Jones [EMAIL PROTECTED] wrote: Connect to the avaya line ports, not station ports. On Jan 18, 2007, at 10:46 AM, Erick Perez wrote: Hi, this is a signalling question: I have a 4port fxs-to-sip where i connect standard analog phones. I want to connect this device to an avaya PBX and then the device talks to asterisk via SIP. What signalling do i need the avaya to provide? FXO signalling right, like this? avayaanalog_lines_fxo_signal--(FXS side)FXStoSIPdevice(SIP side)--Asterisk thanks, -- Erick Perez ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] connecting a FXS-to-sip 4 port device to an avaya system
Connect to the avaya line ports, not station ports. On Jan 18, 2007, at 10:46 AM, Erick Perez wrote: Hi, this is a signalling question: I have a 4port fxs-to-sip where i connect standard analog phones. I want to connect this device to an avaya PBX and then the device talks to asterisk via SIP. What signalling do i need the avaya to provide? FXO signalling right, like this? avayaanalog_lines_fxo_signal--(FXS side)FXStoSIPdevice(SIP side)--Asterisk thanks, -- Erick Perez ___ --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] Caller Id problem
always include a wait before a dial
give the callerid time to get into * before dialing, it arrives
between the first and second ring, if you have * dial after the first
ring it will not be there yet to pass along
On Jan 9, 2007, at 12:16 PM, Anton Frolov wrote:
Dear List,
My problem is that the incoming Caller Id is not displayed on the
local analog
phones (connected to a TDM400 card).
I receive the CID correctly from my telco, but when I place the
call to the
internal analog line, the CID is not propagated.
An interesting point: when I try to place a new call to an already
bridged line,
I see the second call with the CID on the analog phone. The second
call is
placed exactly with the same command/config as the first one.
In the debug log I see (for the second call):
-- Launched AGI Script /usr/share/asterisk/agi-bin/incoming.pl
-- AGI Script Executing Application: (Dial) Options: (Zap/2||otw)
-- CPE supports Call Waiting Caller*ID. Sending '/066332XX'
In other words, the CID is transmitted during a Call Waiting, but
not during a
normal call. It looks like Asterisk does not send the CID (or send
it too soon /
too late) during the first (normal) call.
Any idea is welcome.
Thanks!
AF.
--
*zapata.conf*
usecallerid=yes
usecallingpres=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
treewaycalling=yes
transfer=yes
useincomingcalleridonzaptransfer=yes
...
context=home
signalling=fxo_ks
channel = 1
context=office
signalling=fxo_ks
channel = 2
context=freebox
signalling=fxs_ks
callerid=asreceived
channel = 3
context=francetelecom
signalling=fxs_ks
callerid=asreceived
channel = 4
*extensions.conf*
exten = s,1,Dial(${HOME},,otw)
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Re: [asterisk-users] Any quiet 24 port POE switches out there?
I suspect any 24port will have a fan. The Netgear FSM7326P are not too bad and we have had good luck with them. ps - I also load their open source software. On Jan 3, 2007, at 4:51 PM, John French wrote: I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The Linksys SRW224P behind me right now would be objectionable, I'm sure. ___ --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] PRI ANI/CallerID
add a wait before you dial the sip phone, keep in mind the callerid information arrives later than the call setup info On Dec 31, 2006, at 4:38 PM, David Sampson wrote: For some reason something that seems like it should be simple is leaving me a bit perplexed. I am receiving incoming CallerID ANI on my PRI, but on my VoIP phones the display just shows asterisk when calls come in. I am receiving the calls with DNIS and have the DNIS digits setup as extensions. Do I need to add something to force relay the received caller ID to the phone? Any help is appreciated... 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
Re: [asterisk-users] Remote Reboot of a Polycom
Or web into the phone and click any submit button - not a great idea though if you remotely provision, just make sure you do not change any settings as they will then over ride the remote file settings On Dec 19, 2006, at 1:09 AM, Douglas Garstang wrote: From the Asterisk console: sip notify polycom-check-cfg ipaddr Or you might have to switch the polycom-check-cfg and the ip. I forget the order. You also need to make sure that the phone has alwaysreboot=1 in the sip.cfg xml file. Doug. -Original Message- From: Klaverstyn, David C [mailto:[EMAIL PROTECTED] Sent: Mon 12/18/2006 11:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:[asterisk-users] Remote Reboot of a Polycom Does anyone know how to remotely reboot a PolyCom specifically 601 phone? winmail.dat ___ --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] G.279 license question
OK with the remote server on one side doing G729, what will you be connecting to on the other side? If it does G729 then no license, if not then one license per active call. Also if * will be doing any voicemail etc then you will also need the license. On Dec 19, 2006, at 8:31 AM, Michel wrote: Hi, I need to connect to a remote VOIP server that only uses G.729 codec. From our Asterisk server, we will then make several calls ( 1 but ?? !!) in the same time to the remote VOIP server. Do we need to purchase Asterisk G.279 license ? If yes, how many licenses must we buy? Thanks you! Michel ___ --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] PRI to SIP
Or any of a number of gateways that do this. Off the top of my head you can get one from CarrierAccess, Vega, Audiocodes, Mediatrix, Adtran, and others. Just try to be very careful as they all have their strengths and weaknesses and you need to evaluate how they would fit your needs. Best is to try to get an eval unit and test first - or buy with a 30 day return setup. On Dec 14, 2006, at 6:14 AM, Joao Pereira wrote: For PRI you have 3 main solutions. This is the order of stability (and pricing): 1. Digium or Sangoma cards use the computer processor and that could be bad if you have huge traffic through the PRI 2. Eicon Diva cards have their own processor, which releases the PC processor and gives more stability 3. You can also use a dedicated router (ex: Cisco) to do that.Its more expensive, but more reliable. Regards Joao Pereira Patrick Fortin wrote: Hi Can someone recommend a PRI to SIP Box that work well with asterisk We are presently testing with a Patton Smartnode 2400 but we are unable to fax through it. We don't want to use digium card in a linux box for the PRI connection. Which Cisco box would work. Thanks Patrick ___ --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] RE: Polycom buddies question
Use an empty line key to monitor the other phone On Dec 7, 2006, at 1:44 PM, Bill Gibbs wrote: Figures I email this and realized I can hit Menu 1 (Features) 4 (Presence) 2 (Buddy Status) Wow that’s a lot of key strokes. Anyway to reduce that to a one button touch? I don’t mind doing that but I guess I should think of the users J Bill From: Bill Gibbs Sent: Thursday, December 07, 2006 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Polycom buddies question I know this is not asterisk specific but we all know this group is used for Polycom issues as well… I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip version: 1.6.6.0036 for all platforms except 11402_001 1.6.6.0042 for 11402_001 Any way around this? The big issue is moving from a key system to this is the ability to use the phone to see who is on the phone so you know if you can transfer a call. Obviously web based interfaces work but that draws your attention from the phone to the computer reducing effectiveness. Buddies half solve this… Bill ___ --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] CAS DID 2way
Greetings, I have a customer with an old PBX which cannot accept a PRI. Has anyone tried/tested connecting a CAS T1 to provide 2way DID trunks to a pbx? Either directly to an * server or a gateway? Thanks Jerry ___ --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 Second Incoming Call
you can change the configs to have multiple beeps, and adjust the timing of them, but when we tried the problem then is the beep is not added to the incoming audio, but replaces it, so you lose the far end speaking, went back to default. On Nov 29, 2006, at 3:34 PM, Dovid B wrote: Hi List, I have a Polycom 601 that when the user is on the phone they only hear one beep and the CID of the second incoming call is not shown. Is there a way to have the CID show up for the second call ? And a way to configure the phone to beep more often if there is another call coming in. The problem is that if the receptionist is on the phone and looking up something on the PC she some times dosent realize that a new call is coming in. Thanks. Dovid ___ --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] names of SIP aware firewalls
Intertex Not cheap, licensed per number of users But seem to work great and have some nifty tools very confusing picking models though On Nov 5, 2006, at 3:54 PM, Erick Perez wrote: Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] PRI (TE205P) allways RED/NOP
zttool is your friend here red is LOS or no signal coming in On Oct 26, 2006, at 3:54 AM, Florian Hars wrote: I have a TE205P, jumpered for E1, added the missing wct4xxp-line to /etc/modprobe.d/zaptel, zaptel.conf is just span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 Which, according to my reading of the documentation I could find, should be correct. ztcfg doesn't complain about anything, yet all I ever get is RED NOTOPEN for both spans, and the red blinking pattern on the card changes from alternating to synchronous. Am I missing something obvious? Did I install during the wrong phase of the moon? And what exactly *is* the meaning of the RED alert? Hardware on fire? Configuration Error? No Cable (this would be plausible, I only want to connect it to the ISDN net once the configuration is in a sensible state)? Yours, Florian Hars. ___ --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] RE: ECHO Cancellation in SIP Calls
You will also perceive jitter as echo If any links are getting busy and routers or switches have to buffer you will hear what sounds like echo, not to mention if you have a high packet loss also Of course jitter would have to be above 100ms or so to be noticeable as far as acoustic echo, i have had to put 192ms tail ec on pri direct from carrier because of so many networks interconnecting and doing poor jobs and that 192ms is not going to be enough shortly yes traditionally telco echo originated at 2wire to 4 wire transition points or on hybrids hence it is usually referred to as hybrid echo versus acoustic echo which does happen in an all digital call. This is one thing the better quality phones give you some control of. I am starting to look at dedicated aec hardware to handle even all IP calls On Oct 26, 2006, at 9:56 PM, Michael Araba wrote: I am surprised that you are getting echo on SIP calls. You can get echo in two scenarios on SIP calls. 1. If SIP calls are crossing to PSTN (inbound/outbound). Here you need to enable echo canceller and AGGRESSIVE if needed in zconfig.h. 2. Second source of echo on SIP calls could be ACOUSTIC. The phone sets you are using may not handle this well. In my experience sound quality deteriorates if there is network trouble or congestion on SIP calls I hope this helps. Michael ___ --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] Occasional one-way audio - Sangoma A101
We use almost all Polycoms, several hundred had one way audio with 1.6.4 or 5, forget which 1.66 and 2.01 seem to be ok We did have a few phones (2-3) that had random one way for a long time, replaced everything feeding them and it still happend. A month ago I replaced the phones and have not had a complaint since then. On Oct 19, 2006, at 11:32 AM, Scott Scecina wrote: Hi Mike, Sounds like you're having about the same problem Giorgio and I are having. I'd be really surprised if you don't start having the same problem from SIP-SIP calls to. I also have a Sangoma card, and originally thought it was only on calls coming from a PRI. But as time has moved forward, the issue really appears to be between the Polycoms and Asterisk. The next time it happens, try hitting a digit (like 5) on the polycom and see if the audio becomes available. BTW - our other discussion on this is called random one way audio and noise between SIP phoneson same LAN - Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Thursday, October 19, 2006 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Occasional one-way audio - Sangoma A101 We are having an occasional one w-way audio problem that occurs about every 25 - 30 calls on a system configured as follows: Asterisk 1.2.12.1 Sangoma A101 w/wanpipe beta9 Polycom 500s w 1.5.3 This happens only on inbound calls from the PRI. The external caller can hear our customer answer and say hello, however, our customer cannot here their caller. Typically, the caller calls right back and all is fine. There is no discernable pattern as I can tell. Anyone have, and hopefully fix, a similar issue? Thanks, Mike Clark ___ --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] Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)
Sounds like they are rebooting. Is power being interrupted at night? On Oct 13, 2006, at 9:40 AM, Mike Garey wrote: I've been noticing that my group of Polycom IP 501 phones seems to randomly reset themselves nearly every night (I guess it usually happens at night, since I've never seen it happen while I've been at work during the day).. When I say reset, I mean, the hands free volume and ring volume are set to the default and the call logs (received calls, missed calls, placed calls) are all reset. It does, however, keep certain settings such as the specific ring tone used for incoming calls.. But most other settings are being reset.. Has anyone else experienced this, or know why it might be happening? 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 ___ --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 HDVoice
Resellers claim it will ship in December or there abouts Uses g.722 About $30 more than 601 On Oct 13, 2006, at 11:14 AM, Forrest Beck wrote: Has anyone used the Polycom HDvoice phone yet? I am curious if it uses a different codec. Does it actually sound any better? ___ --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] Polycom 601 Expansion Module: Light the LEDs???
Enable buddy watch in your poly config files also set each speed dial to have this enabled also On Oct 9, 2006, at 12:04 AM, Doug wrote: Hey Folks, Been wrestling with the 601 and the expansion module. Finally figured out how to populate both with speed dial entries. Also hints are showing in Asterisk with the show hints command. But how do I get the LEDs to light when one of these other extensions is either off-hook, or ringing. Reading the 'Net and Polycom's documentation doesn't give a clear solution. Is there a genius out there who has this working?? Please help!!! ___ --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] DSL router with integrated SIP proxy?
We are trying a couple of the Intertex - seems to work so far On Sep 29, 2006, at 2:59 PM, Andrew Joakimsen wrote: The VoIP version of DD_WRT runs Ser by default On 9/24/06, David Gagnon [EMAIL PROTECTED] wrote: You could take a WRTSL54gs, install openwrt then openser David -Message d'origine- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Steve Kennedy Envoyé: 24 septembre 2006 08:47 À: asterisk-users@lists.digium.com Objet: Re: [asterisk-users] DSL router with integrated SIP proxy? On Thu, Sep 21, 2006 at 10:11:43PM +0100, Brian Candler wrote: Does anyone here know of an ADSL router with integrated SIP proxy? Netscreen 5GT ADSL, it has what's called an ALG (application layer gateway) and it does indeed support SIP. Full featured firewall etc too. Steve p.s Hi Brian :) -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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 ___ --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
[asterisk-users] Polycom (and others) digitmap info
There have been many threads regarding specific uses for digitmaps. One of the most common is for the telephone to perform digit substitution and prepend some digits. Never thought this was possible until I found a reference in a Sipura tech note . Anyway hope this helps someone. Add something like :9xx to your digitmap and it will prepend a 9 to the 10 digit number. Tried on my Polycom phone and works great for just hitting dial when reviewing missed calls. Did have a pattern match in my dialplan but it created other issues and was removing it. This may actually work better. ___ --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 2.0.1 Software
Had problems the first night I downloaded and installed, but tracked to very poor net conditions. Reloaded this week and all has been working fine. Nice to finally be able to use all the buttons on the sidecar for blf:) It may be my imagination, but it also seems that it is staying in sync through reloads, or at least resyncing shrtly after one. On Sep 20, 2006, at 10:13 PM, Douglas Garstang wrote: No problems with SIP subscriptions here... -Original Message- From: Lacy Moore - Aspendora [mailto:[EMAIL PROTECTED] Sent: Wed 9/20/2006 8:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Polycom 2.0.1 Software I couldn't get the hinting to work. Went back to 1.6.7, same config, and it works. I wasn't sure if the config had changed between the two. But, now that you mention it, I did experience a phone rebooting several times. I was half-way paying attention, so I just thought I had done something. On 9/20/06, Douglas Garstang [EMAIL PROTECTED] wrote: Is anyone seeing any weird stuff with the latest Polycom 2.0.1 SIP application software? A few of our phones, after upgrading would come up with a 0x4000 Configuration Error. Rebooting again a couple of times, or doing a 'Format Local Filesystem' seemed to fix it, with no change to the config files on the FTP server. I've also had an instance where a phone was refusing to register after upgrading. It worked fine, first boot, after doing a 'format local filesystem' on the phone, Doug. ___ --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 -- Lacy Moore Aspendora, Inc. ___ --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] Re: Two phones, same number
set group/check group On Sep 21, 2006, at 8:22 AM, Benny Amorsen wrote: ZZ == Zeeshan Zakaria [EMAIL PROTECTED] writes: ZZ Why don't you simply give them separate extensions and put them in ZZ a ring group. I'm not quite sure what you mean by ring group. Perhaps you could elaborate? ZZ Or disable call waiting on this phone, and forward the second call ZZ using Call Forward On Busy to a queue, where MoH file will be a ZZ busy phone signal. Called will hear a busy phone signal and the ZZ second phone will still be ringing. I don't want the second phone to ring. ZZ But whats the point to make the second phone ring if caller is ZZ hearing a busy tone. He'll hang up anyways. I want the caller to get the busy tone. Basically, if I'm talking on one phone, I don't want the other phone to ring. /Benny ___ --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] Polycom Expansion Module
Per 2.0.1 release notes 13315: Increased the maximum number of buddies to 8 for all platforms except SoundPoint IP 600 and 601 which can watch 48 buddies On Sep 18, 2006, at 10:35 PM, Douglas Garstang wrote: 48 was the limit on the number of speed dial entries that you could have in the directory. 7 was the old limit for the number of buddies you could watch. As far as I know, in 2.0.1, the number of entries you can have in the speed dial directory is 99, and the number of buddies that you can watch has gone up to 48. Doug. -Original Message- From: Jerry Jones [mailto:[EMAIL PROTECTED] Sent: Mon 9/18/2006 6:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Polycom Expansion Module Poly 2.0.1 says it can do 48 On Sep 17, 2006, at 8:06 PM, Douglas Garstang wrote: As far as I know, it's 12. -Original Message- From: Noah Miller [mailto:[EMAIL PROTECTED] Sent: Sun 9/17/2006 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Polycom Expansion Module Hi Kevin - Has anyone used the Polycom expansion module with multiple lines? My application is for 20 lines and read there was a limit of 7 at one point. I heard rumors that the newest version of the polycom sip firmware (2.01) would lift the limit of 7. It just came out, and I haven't had time to test it yet, but you can give it a try. - Noah ___ --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 ___ --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] polycom 501 digitmap
the digitmap only tells the phone when to send the digits it has collected. They have no digit substitution feature. This would be done within your * dialplan On Sep 19, 2006, at 7:57 AM, Jordan Novak wrote: This is really starting to get to me. I have deleted this field in the phones per the wiki. I am trying to get the phones to dial on there own. Is there anyway to get the phone to dial 1-8 after three digits are received and 9 after seven to ten digits. I am willing to wait for a timeout but that doesn't seem to work. Any help is greatly appreciated. Jordan Novak Senior Telecommunications Engineer Logistics Health Inc. ___ --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] Aastra 9133i and Atcom AT-320 - Comments please
We tested a couple 9133i, dont remember the specifics right now but we stopped as there was some inconsistency in provisioning. I was very optimistic as I like the look and feel. We did deploy a couple 480iCT which worked very well - when they worked. But they keep locking up and freezzing under heavy use, plus they have speakerphone and rfi issues. You may wish to checkout the Polycom IP430, about the same price and have been very solid after installing 50 or so to date. On Sep 19, 2006, at 8:15 AM, Dave Cotton wrote: On Tue, 2006-09-19 at 13:13 +0100, James Dyer wrote: I'm planning to deploy an Asterisk system in our office soon, and am thinking of using a mixture of Aastra 9133i and Atcom AT-320 phones. Has anyone got any comments (good or bad) about these phone models? I now only use Aastra phones, the 9133i is solid and professional looking and works very well with *. My experience with support is A1. Message waiting is well signalled as is no service. The switch and POE save a lot of cabling. -- Dave Cotton [EMAIL PROTECTED] ___ --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] Polycom Expansion Module
Poly 2.0.1 says it can do 48 On Sep 17, 2006, at 8:06 PM, Douglas Garstang wrote: As far as I know, it's 12. -Original Message- From: Noah Miller [mailto:[EMAIL PROTECTED] Sent: Sun 9/17/2006 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Polycom Expansion Module Hi Kevin - Has anyone used the Polycom expansion module with multiple lines? My application is for 20 lines and read there was a limit of 7 at one point. I heard rumors that the newest version of the polycom sip firmware (2.01) would lift the limit of 7. It just came out, and I haven't had time to test it yet, but you can give it a try. - Noah ___ --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] Hardware ? Analog DID trunks (ILT)
Do not know of a card that does. But think a digium T1 to a channel bank (ie Adit600) would. On Sep 1, 2006, at 2:06 PM, Tim Sharp wrote: I am looking at CTPX's VP2000 product. I haven't tried it yet. Please let me know if you find a solution that works. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] Behalf Of Jonn R Taylor Sent: Friday, September 01, 2006 12:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hardware ? Analog DID trunks (ILT) Is there a card that supports analog DID trunks, alosi known as ILT trunks or Incoming Loop Trunk. They work by providing talk battery to the CO, incoming calls happen by pulling loop sending a wink accepting the DID dtmf digits for the station being called. Jonn ___ --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] Re: Adit 3104 randomly reboot
We used some way back (a year ago) when they first came out. Had several issues which they were very helpful in working with us on. They resolved many, had to upgrade and load patches. Unfortunately they were lacking a couple features we required so they have been replaced. Give tech support a call, they will help. On Sep 1, 2006, at 1:04 AM, Martin Joseph wrote: On 2006-08-31 19:12:03 -0700, Xue Liangliang [EMAIL PROTECTED] said: Hi, all. I have a Adit 3104, and I configured it to work with Asterisk, the voice quality is quite good, however it just randomly restart, I don't know whether you guys have the same experience, is it due the firmware bug? I don't know that hardware at all, but it kind of sounds like a hardware issue (ie power supply?). Just guessing. Marty ___ --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] Polycom 501 config questions
On Aug 30, 2006, at 2:58 PM, Mike wrote: Hi, I have a few questions on the Polycom 501. I am using latest firmware. 1) When I press the Call List button (on the left row of buttons), I get the call lists (as expected). When I press the Directory button, I get the choice between Directory and Call lists. How can I make this button go to Directory immediately? 2) I have 2 extensions on my 501. (let's say 101 and 102). Because of my dialplan, it actually matters which one I dial out with. When I pick a contact out of the directory, it calls automatically using line 101. How can I make it call with 102? Pick up 102, then select contact 3) In call lists, my numbers are listed as 555-555-. Yet my asterisk dial plan requires me (by design) to press 9 first. How can I make the phone put the 9 by itself? It will not. either add to your contact entries, or alternatively have your dial plan add 9 to any exten longer than say 3 digits Thank you for any help you may give me, 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 ___ --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 - SIP client latency
Such an objective question. Everyone, including different users will have a different answer. Is this within an enterprise? at home? with a paid service? what codec? pure IP or TDM mix? I would say anything over 200 is bad, now how close you get to that. We try to engineer our on net to sub 100 of course our echo cans tell us the PRI to the PSTN regularly hit over 150ms which is ridiculous, and keep getting worse On Aug 19, 2006, at 12:04 AM, Freddy Setiawan wrote: Heya all, what is the acceptable latency for VoIP calling? 200ms? 300ms? Best Regards, Freddy Setiawan ___ --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] Polycom upgrade issue
Manually config to point to your boot server, which should have a good copy of the software and it should go get it. If not sniff the traffic in/out and see what it IS doing. I have had several firmware updates get interrupted in the past corrupting the image and this has always worked. On Aug 16, 2006, at 12:15 AM, Dovid Bender wrote: I believe 468* resets the phone but dosent return it to the orig. firmware. Also try to name the files with the phones mac id and see what happens. I am doing this with 1.6.6 and its working fine. - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, August 15, 2006 10:07 PM Subject: [asterisk-users] Polycom upgrade issue OK, I may have done something stupid. I was trying to upgrade my Polycom to the newest firmware I could find (1.6.7). I am also trying to get provisioning working from a central server. I tired to reset with holding 468* down and it kept the settings the phone had on the phone. From what I understand the settings on the phone override all. So I went into reset it from the phone and choose to format the firmware. Now when I try to boot it I am getting the following in the *-boot.log 0527180621|cfg |4|00|Could not get all 512 bytes of the header. 0527181013|cfg |4|00|Could not get all 512 bytes of the header. 0527181014|app1 |6|00|Error application is not present. 0527181014|app1 |6|00|Uploading boot log, time is SAT MAY 27 18:10:14 2006 I tried to put the old firmware and configs back in the directory but I get the same thing. Any help out there? 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 ___ --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] What to use beyond T1's?
rumor has it Sangoma will be releasing their ct3 card in a couple months do no transcoding or EC and one server can handle a large quantity of T1s On Aug 16, 2006, at 8:52 PM, Matt Florell wrote: Use multiple servers. What kind of calls are you handling that you can have more than 3 quad T1 cards in a single server and have your Asterisk-based application be functional? And what kind of server hardware are you using? MATT--- On 8/16/06, Steve Edwards [EMAIL PROTECTED] wrote: When you get beyond a dozen T1's or so what have you done? We've configured t1 servers with quad T1 cards which hand off calls to application servers via IAX and this is working pretty good, but, where do we go from here? Talk about Digium's DS3 card appears to have evaporated. What about a Tekelec or a Max TNT? What have you used to get in the neighborhood of 1,000 simultaneous calls into your Asterisk applications? How much did your solution cost and what problems did you experience? Did you drag a DS3 to your location or did you co-locate at the CO? Inquiring minds want to know :) Thanks in advance, - --- Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] Prevent a Polycom contact list to be overwritten
Been awhile but IF memory serves... Manually enter the boot server IP on the phone. I do not think causes a reboot - of course this was several versions back in sofware. Then edit a contact and press save. Every time it updates the list on the phone, it tries to copy to the boot server. This should create a new xml contact file for you. Then just go ahead and reprovision from the server. On Aug 3, 2006, at 6:43 PM, Stephen Murphy wrote: That’s exactly what I want to do – download the xml file from the phone any ideas? From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Jim Freeze Sent: August 3, 2006 4:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Prevent a Polycom contact list to be overwritten On 8/3/06, Stephen Murphy [EMAIL PROTECTED] wrote: I have a Polycom phone that was setup without provisioning through an FTP server. It has a number of contacts that where input via the phone. I would like to add this phone to a small network that was provisioned through an FTP server and keep the contacts already on the phone. How do I ensure that the contacts list file will not be overwritten when I do a provisioning? I would like to know this as well, but for a slightly different reason. I want to provision 501 phones, but I want to start from what is currently on the phone. So, in other words, I want to download the XML file that is stored in the phone. Anyone know how to do this? Thanks -- Jim Freeze ___ --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] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems
If you see no errors on your MX2800 for the ds3 then they are probably not the issue. What does the MX2800 show for T1 which do not work? If you loop toward * does the card see itself? Loop toward GX do they see? On Aug 4, 2006, at 11:15 AM, Steve Totaro wrote: I have a DS3/T3 that was dropped into my telco closet as two coaxial cables. A send and receive. I needed to extend it so I went to Radio Shack to buy some barrel connectors. They did not have any but they did have T-Connectors so I bought a couple of those. Everything was fine up until I went to turn up the seventh set of four T1s. They will not come up, no Asterisk output or greenlight on a Digium TE410. Global Crossing says they only responsible for the DS3. I wonder if the T-Connectors could be the problem? I have already tried the same cables and server on known working T1s and they come up fine. My Adtran MX2800 shows OK for those T1s so I dont think the problem is between the Adtran and Asterisk box (as Global Crossing claims), I have tried changing to the second controller card in the Adtran and also reseated the amphenol connection between the Adtran MX2800 and the T1 breakout box. Anyone with this kind of experience have any ideas? I am going to find some real barrel connectors and try that next. Thanks, Steve ___ --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] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems
The MUX will give you stats for every line running through it, DS1, DS2, and DS3. Start there. What errors is it reporting? Actually I did that backwards, start with DS3 then do DS1, if 3 has issues so will everything else. On Aug 4, 2006, at 1:04 PM, Steve Totaro wrote: I am not totally up to speed on the MX2800 but I have gone to loopback tests and loop T1 25-28 and selected every possible selection while watching pri debug span 1 on the console, no output at all. Jerry Jones wrote: If you see no errors on your MX2800 for the ds3 then they are probably not the issue. What does the MX2800 show for T1 which do not work? If you loop toward * does the card see itself? Loop toward GX do they see? On Aug 4, 2006, at 11:15 AM, Steve Totaro wrote: I have a DS3/T3 that was dropped into my telco closet as two coaxial cables. A send and receive. I needed to extend it so I went to Radio Shack to buy some barrel connectors. They did not have any but they did have T-Connectors so I bought a couple of those. Everything was fine up until I went to turn up the seventh set of four T1s. They will not come up, no Asterisk output or greenlight on a Digium TE410. Global Crossing says they only responsible for the DS3. I wonder if the T-Connectors could be the problem? I have already tried the same cables and server on known working T1s and they come up fine. My Adtran MX2800 shows OK for those T1s so I dont think the problem is between the Adtran and Asterisk box (as Global Crossing claims), I have tried changing to the second controller card in the Adtran and also reseated the amphenol connection between the Adtran MX2800 and the T1 breakout box. Anyone with this kind of experience have any ideas? I am going to find some real barrel connectors and try that next. Thanks, Steve ___ --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 ___ --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 and Siemens Legacy PBX
probably need a crossed t1 cable 1-4 2-5 On Aug 4, 2006, at 4:20 PM, James Arscott wrote: Hi, this is my first post, so go easy on me ! Sorry if this has been covered before, I could not find an answer that helped me. I am trying to achieve the following : Telco ISDN30e PRI - Asterisk with TE210P - Siemens HiPath PBX The siemens is a legacy PBX and I am not 100% of the modules etc inside it, it is being used in production at the moment and we have a need to put the Asterisk pbx as a gateway in between the ISDN and the Siemens. Ultimately this will help us move people from the legacy PBX to full SIP phones. We have many Asterisk PBX's working well using the TE210P + ISDN30e PRI, but I am unsure how to get the legacy PBX working with the 2nd span of the TE210P. I *assumed* that all I had to do was configure the 2nd span with pri_net and leave span 1 as pri_cpe and that would do the job, but when I do this and plug the siemens into span 2 I get a RED alarm on the span 2 and that's about it. Any tips on the most likely configuration that will work ? What configuration of CAT5 should I be using to connect the legacy PBX to span 2 ? Straight, crossed, etc. Many thanks in advanced ! James ___ --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] Polycom 501 : How to make it ring when alreadyona call
You will get a call waiting beep. One. However you can change the config file and have multiple beeps. You can also change the beep 'sound'. However you must also be aware that while the phone is playing the beep(s), you are not hearing the far end of the call. On Aug 3, 2006, at 9:21 AM, Mike wrote: Thanks, I know your right (I tried the second option). Problem is that the phone doesn`t RING. The light flashes, the as far as an audio ring goes, it`s completely silent. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Vincent (C) Sent: August 2, 2006 9:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom 501 : How to make it ring when alreadyona call So I`m throwing this question back in the arena: Can you get the Polycom 501 to ring when a calls comes in and the user is already on a call? Two ways: 1. Call forward the first extension in a busy condition to another extension registered on the phone. I use trixbox and with that it's a *90 and then the extension to forward to. (On the second line, send busy and don't answer to voicemail.) 2. You can have multiple calls to the same SIP registration using the Polycom config file. I believe the parameter in the extension registration is something like callsPerLineKey=X where X is the number of calls to get to that extension. --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] __ __ __ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank 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 ___ --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] long distance ethernet Asterisk
It has been several years since I had to address similar situations, but I used TUT Systems devices back then. worked great. There are several DSL variants which should work ok. On Jul 27, 2006, at 6:02 PM, Manrique Feoli wrote: another thought, if you are in a bowl, all you need to find is line of sight to one common place from both ends, and place a repeater there. (you could also set two or three steps repeating the signal within points which have line of sight). I'm not sure but I think one repeater would be much cheaper than 20.000ft of copper + extenders + poles+ maintenance, lighning... (even thought you are in Copper Mountain !!, BTW nice spot ). if in the end you decide to go with ethernet, just beware of lighning!!! Brian Vincent (C) escribió: I know.. I know… fiber would be ideal. We have single-mode all over the place. We even have some dark, unterminated strands within 2000ft of this location – it makes me want to cry. Unfortunately lighting it up isn’t an option – we wouldn’t gain anything because we couldn’t connect to anything else to get us the last stretch. Trenching 2000ft isn’t an option – this is National Forest land and we’re not allowed to do that. As far as wireless – no line of sight. This location sits in a little bowl at 11,200’. So what I’m left with is a 400pr, 22awg out to 3000’. Then we jump on 200pr, 24awg aerial cable strung on the 3^rd longest high-speed quad chairlift (10,800’ run). The last leg involves a short underground to another high-speed quad and down 6000’. We can stick a powered repeater in the motor room of the first lift (so I guess a bit further than the original 12,000’ I was thinking.) Yes, we do strange things. If you’re really curious, here’s a map of the campus environment we maintain: http://www.skireport.com/colorado/copper/trailmap/ --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] -Original Message- *From:* [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] *On Behalf Of *Bruce Reeves *Sent:* Thursday, July 27, 2006 4:03 PM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] long distance ethernet Asterisk I would really look towards fiber, the bandwidth and distance can easily be handled. On 7/27/06, *Manrique Feoli* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If you have line of sight between the points, maybe you could setup a wireless link point to point, I know some people who have done it over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you could get more). just a thought Joe Pukepail escribió: Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, *Brian Vincent (C)* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Two questions: 1. We need to run Ethernet out to a really long distance – 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 – 10Mbps. 2. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] _ _ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. _ _ ___ --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 - --- ___ --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
Re: [asterisk-users] Just bought a Polycom 501 - I feel likemyGXP-2000 was better...
I really like the IP60x phones. Have started using the IP430, so far after 20 or so they are fine. But the IP30x and 50x I refuse to use. The aastra 480i is also good. The 9133i has promise. I do not like the snoms - any. Grandstream are so so Budgetone is not bad for the price, but not enterprise grade. My evals are based on useability, quality, reliability, and management. pros and cons on all, but the 601/430 are my best picks so far. I have tested and used Cisco also, but their price and license and feature models are nuts, at least the last time I really investigated. On Jul 26, 2006, at 12:27 PM, [EMAIL PROTECTED] wrote: On Mon, 24 Jul 2006, Douglas Garstang wrote: Not for our users. We held focus groups, and the Polycom's won in terms of ease-of-use over all the other phones investigated. Which other phones did you investigate specifically? Our users found the polycom menus cumbersome, with commonly used options buried 3 or more levels deep. Transfers don't work the way users expect (blind vs attended), and other issues. -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 ___ --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
