Re: [asterisk-users] No music on hold?
Heya, > I've got Asterisk 1.2.10 up and running on Debian using the back ports. > I noticed that it didn't come with mpg123 or depend on it and I believe > I read somewhere that asterisk now handles it's own mp3 playback? Is > this true? If so I must have a problem, because I hear no music when > putting someone on hold. When looking at the console when putting > someone on hold, I see the following: > > -- Started music on hold, class 'default', on channel > 'IAX2/voicepulse01-3' > -- Stopped music on hold on IAX2/voicepulse01-3 > > It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. J. ___ --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] No music on hold?
> On Tue, Dec 19, 2006 at 10:04:23PM -0500, Jerry wrote: >> Heya, >> >> > I've got Asterisk 1.2.10 up and running on Debian using the back >> ports. >> > I noticed that it didn't come with mpg123 or depend on it and I >> believe >> > I read somewhere that asterisk now handles it's own mp3 playback? Is >> > this true? If so I must have a problem, because I hear no music when >> > putting someone on hold. When looking at the console when putting >> > someone on hold, I see the following: >> > >> > -- Started music on hold, class 'default', on channel >> > 'IAX2/voicepulse01-3' >> > -- Stopped music on hold on IAX2/voicepulse01-3 >> > >> > It says music starts and then it instantly stops. Any ideas? >> >> Do you have asterisk-addons installed? That could be the issue. > > Why? The OP is looking to play MP3s, and unless I misunderstood the instructions on the Wiki, addons is required (format_mp3) to play MP3's on 1.2.x. Is that not the case? J. ___ --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] No music on hold?
Hi Phil, > No, I didn't have m added. Should I have it added? I know I've ran > Asterisk with mp3123 in the past and music worked ok. It seems when I > hit the hold button on the phones, it does trigger the message saying > music on hold is starting but it INSTANTLY stops. I wish it gave some > details as to WHY it stops suddenly! This is driving me nuts. Do you see the mp3 format in your list of modules? (do a "show modules" from the CLI) I think this was what I had when the module wasn't loaded for me. The m option to dialing plays music instead of ringing while dialing, which (from my understanding) isn't what you were after. J. ___ --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] No music on hold?
> On Wed, Dec 20, 2006 at 08:30:27AM -0800, Kevin Trumbull wrote: >> I already posted about this, but contrary to what is stated on the Wiki, >> mpg123 is required (at least in 1.2.x) if you wish to use mp3's for your >> MoH. >> >> I decided to go this route: >> http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it Kevin - thanks for that link. We're now using native slin for music on hold, but it's a good resource. And thanks for pointing out the Wiki is actually wrong on this. :) > If you're not streaming the MP# from an external source, converting it > off-line will always be cheaper. And it may even actually save you disk > space, because mp3 files have a much higher quality than Asterisk > requires. Tzafir - both true, but I believe the OP was trying to play MP3's. It's kind of silly that Asterisk includes mp3 files, and then there isn't a way to play them. (There could be licence reasons for this, or another reason I'm not aware of, but people will assume that MP3's work.) And, there is a certain coolness factor in it "just working", of course. Does format_mp3 only work properly in 1.4 then? I'll dig a bit and correct the wiki, if I can find a definitive answer. (I thought it was working on 1.2, but I've installed a few 1.4 boxes recently for testing, so my recollection could be of a 1.4 install). Thanks, J. ___ --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 + door opener
Hi Dovid, > I am actually now working on massproducing door > openers that will work with asterisk. It will have an > rj45 port and then a port to plug the door opener in > to. Please contact me off list if you are interested. This is an old message, but I was wondering if you are still doing this, and what the specs/cost are. Thanks, J. ___ --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] Escalate Call To Mobile
> Thanks, this it seems was what the problem was. Is it possible to
> specify number of rings instead of timeout in seconds?
Short answer: no.
Longer answer: kind of.
As alluded to here:
> Also the timeout to Dial is in SECONDS, not RINGS. In the USA a ring
> cycle is about 6 seconds.
You can approximate how many rings have happened. But, the indication that
is sent to you (the ring you hear when you are calling someone) isn't
synced with ringing of the target phone (which has to do with the far end
ring generator, if I recall right), there is no way to be sure.
So, at the end of six seconds of "ringing", you are sure that the phone
has rung at least once. It might not have started until late in the cycle,
or you might have caught the tail end of one cycle and it's already
starting to ring a second time.
So, if you wanted to make sure it rings 5 times (in the USA), you have to
let it ring for a minimum of 30 seconds.
If it's easier, you could set this up as a variable called RINGCYCLE and
just multiply your rings by it, like so:
exten => _NXXNXX,1,SetVar(RINGCYCLE=6)
exten => _NXXNXX,2,Dial(Sip/Provider/{EXTEN},$[${RINGCYCLE}*2])
(I think you could make RINGCYCLE a global var, but check to be sure).
Hope that helps,
J.
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Re: [asterisk-users] Some queries on g729 license.
>> What about the free open source G729 > > There's no such thing ... g.729 (as per the ITU specification) is patent > encumbered. Anyone USING the codec has to pay a license to the patent > holders. I believe (this may have changed) that ANY patented technology can be used for free educationally. The idea is that people can study and play with the technology for no charge. I'm not sure if this means that a University can use this in their phone system without paying the patent fees, though. Now, certainly there can be "open source" versions of the G.729 codec. They can even be "free" in the sense that the author is not charging. But the author can't waive the patent rights. Intel has a freely downloadable codec for educational use, but they have a long legalese document which explains the patent obligations. If you are using G.729 commercially, there is no question you have a legal obligation to pay the patent holder for his rights. J ___ --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] Some queries on g729 license.
> Jerry wrote: > >>I believe (this may have changed) that ANY patented technology can be >> used >>for free educationally. The idea is that people can study and play with >>the technology for no charge. I'm not sure if this means that a >>University can use this in their phone system without paying the patent fees, >> though. >> >>Now, certainly there can be "open source" versions of the G.729 codec. >>They can even be "free" in the sense that the author is not charging. But >>the author can't waive the patent rights. >> >>Intel has a freely downloadable codec for educational use, but they have >>a long legalese document which explains the patent obligations. >> >>If you are using G.729 commercially, there is no question you have a >>legal obligation to pay the patent holder for his rights. > Whether it's a university or a megacorporation studying the technology, > they have to be very careful. > > Suppose we are working on automotive fuel economy or emissions > improvement. If we buy a new or used car we are reasonably sure that a > multitude of patents involved are being legally used. If we build a > cadillac clone for the research, I would be worried. I can't find any refs to it now, so it might well be that the law has changed. It was very specifically for "research purposes only", and may have been limited to educational institutions. I'll dig around and see if I can locate the reference. > Another factor to consider in some cases is when we sign a sales > contract that includes things like "no reverse engineering". It might be > hard to prove that we did not reverse engineer the product in order to > develop a patentable improvement. Since patents require the disclosure of the novel process in order to be considered, reverse engineering or violating the DMCA (or whatever other law may apply) wouldn't have to be done. That's the beauty of patents, as opposed to keeping something a trade secret -- anyone can see what is being done. (Reverse engineering is generally done to inter operate, not to recover something patentable) J. ___ --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] NAT solutions
> On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote: >> >> Unless you are monitoring calls, want full CDR etc, >> then that's what you want anyway. > > CDR are not affected by how the audio flows. While technically true, I believe (it may have changed in 1.4) that if you allow reinvites, the signalling path follows the audio path, and you end up with reported calls lasting 3 seconds. So, if you want full (ie accurate as to the length of time) CDR, then I think asterisk has to remain in the call path. J. ___ --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] Configuring the Sipura for static IP and registering with Asterisk.
OK so now you have an IP address. Did you login and configure the Sipura? On Apr 7, 2005, at 1:04 AM, Rich Adamson wrote: I wish to configure my Sipura with static IP. I have set the static IP, but there is registration failure on doing so. Could you please tell me how do I go about configuring my Sipura for static IP and register it successfully with the Asterisk server. A few of the spa changes require the box be rebooted. Did you do that? Can you ping the sipura's ip address? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank replacement
I enjoy using the Adit 600 with the new FXS cards via the controller T1 interfaces. Works well. I do have concerns with using the CMG card via MGCP. Has anyone done this? How is it working? On Apr 8, 2005, at 12:50 PM, Matt Schulte wrote: Word of warning, get the version 5 or higher FXS cards with the ADIT600, else you will have echo problems. This is just from personal experience. Supposedly the 5 and higher cards have dynamic impedance adjustment, it's worth it. Matt -Original Message- From: Peter Hoppe [mailto:[EMAIL PROTECTED] Sent: Friday, April 08, 2005 12:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Channel bank replacement Thank you so much for your answers already, I really appreciate it! I have looked into using an Adtran Total Access 750 platform instead, but got away from that idea after I saw the totally confusing amount of options of different modules I can buy. The Adit 600 seemed so much simpler to put together. Also, the Adit 600 had such an excellent appraisal in the asterisk voip-info - see http://www.voip-info.org/tiki-index.php?page=Asterisk%20Channel%20Bank But maybe I need to come back to the Adtran TA750. Unfortunately that platform seems to only offer 24 fxs ports per unit and I need to buy an expensive T1 card. I would buy the Digium T1 card - it seems that it is by far the least expensive card, but $500 is still something. That's why I toyed with the Adit 600 plus cmg card - all I need is a standard network card on the Asterisk machine. We have sorely abandoned the idea of using an extensive amount of voip phones on the property, as we are not a homogenous office setup (ppl also live on the property). This solution would mean * putting in an entire new cat5 network. I would be the person who would have to put it all in place - When would I be finished? In 2 years? 4 years? 10 years? * lots of admin hassle to enable all the phones / add new phones / remove phones * users can't easily extend stations at end points. With two wire phone they simply switch one parallel to the existing one - no admin hassle / extra hubs etc. * two wire technology enables us to buy almost any phone available. * security concerns with the SIP protocol. See http://secunia.com/advisories/8169/ as an example * users potentially plugging their laptops into the voip sockets and browsing/downloading away => lots of setup/admin hassle with the firewall (how do you block Kazaa?) * Phones potentially breaking when users unplug power during firmware download. For example, this is an issue with the Grandstream phone. The only alternative that seems feasible at the moment would be * a different channel bank than the adit 600 or * a voip gateway that multiplexes many fxs ports into one ethernet connection. But before I would go down that route I would have to be absolutely sure that the SIP conforms to the standard, the upgrades are free and the fxs ports are compatible with uk standard two wire phones. I found that some two wire phones actually use 4 wires - confusing * a bank of ATAs (handytone 286 or similar). I *really* don't like that solution, as it is a bad botch job and throws lots of issues like which REN they have, many power supplies (or one big one). I really ought to be red in the face for even mentioning that solution. But if nothing else is available, I would probably have to buy them in bulk, take the boards out and mount them in a 19'' box together with a hub so I build my own voip gateway :) maybe it's not so botch after all :) ) For connection to the PSTN: We have three BT lines, and again, we would not like to move over to a different technology like ISDN. The lines work for us, and 'if it ain't broke, don't fix it'. We would use three Sipura SPA-3000 interfaces to connect them to the internal network. The SPA-3000 is sold in the UK and has the CE approval, so it should legally be ok. I am experimenting with one unit at the moment, and am smacked by the literally hundreds of options it has. But I heard good reports about that one, so I expect it to work well in our setting. Hi Peter, I'm not sure how you are getting PSTN lines into your * box, but if it's not ISDN30, you might want to consider some of the cheap IAX phones on the market now rather than trying to soldier on with old analogue kit? e.g. http://www.iaxtalk.com/product_info.php?cPath=1&products_id=29 Shipping for 30 units and UK power supplies was $340, and with the weak dollar right now, that works out at just over 40 quid per phone - I'm sure there's movement on the unit price when buying in bulk... Now remove the need for an Asterisk Quad-E1 / T1 interface card and you've dropped the cost by nearly a grand food for thought :) They also sell a single-ethernet-port version of the phone for $10 less if you have enough ethernet sockets. Cheers, Gavin. I got an Adtran 600 with 12 X FXO and 12 X FXS cards for $495 from Penny Doyen [EMAIL PROTECTED] With the strength of the poun
Re: [Asterisk-Users] multiple line usage on Polycom IP300
Polycom enables call waiting on each line button. If you wish the second call to go directly to the second button you need o keep track of this with group in * and control with your dial plan. On Apr 12, 2005, at 9:41 AM, Josiah Bryan wrote: On Tuesday 12 April 2005 10:18 am, MobilPete wrote: can anyone help ?? trying to get Polycom IP300 to utilize both lines, would like calls to roll to open line when incoming call arrives while user is on line 1. Looked everywhere and tried many things with no luck. Do you have your lines register sepratly? E.g. is there a seperate entry in sip.conf for each line or do they both register as the same sip device? -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat doesn't work
Try adding a comma to your digitmap where you wish the dialtone to come back on. Works on a Polycom. On Apr 23, 2005, at 7:12 PM, Eric Wieling aka ManxPower wrote: Grandstream does not support a dialplan. It is supposed to support Early Dial, but didn't work. I've been told that recent firmware fixes the early dial bug. I doubt that Early Dial is the solution. The solution is to buy a good IP Phone. Polycom and SIPura both support "continue dialtone after digit". Cisco ATAs do not. I don't know if the Cisco IP phones do or not. Alexander Lopez wrote: ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that "ignorepat" only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the "ignorepat" in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat doesn't work
The digitmap is in your telephone. Used to terminate dialing and send the dialed string to *. On Apr 23, 2005, at 11:56 PM, Jaime Blanco wrote: Jerry, when you say digitmap, you mean in my extensions.conf file? Thanks. Jaime From: Jerry <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ignorepat doesn't work Date: Sat, 23 Apr 2005 19:44:20 -0500 Try adding a comma to your digitmap where you wish the dialtone to come back on. Works on a Polycom. On Apr 23, 2005, at 7:12 PM, Eric Wieling aka ManxPower wrote: Grandstream does not support a dialplan. It is supposed to support Early Dial, but didn't work. I've been told that recent firmware fixes the early dial bug. I doubt that Early Dial is the solution. The solution is to buy a good IP Phone. Polycom and SIPura both support "continue dialtone after digit". Cisco ATAs do not. I don't know if the Cisco IP phones do or not. Alexander Lopez wrote: ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that "ignorepat" only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the "ignorepat" in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interrupting voicemail with "*", dropping to "a" extension. Does it work?
Just got done testing this:) * drops into a for me every time on 1.0.7 Pressing 0 does nothing however and that is what my customers wish to use On May 12, 2005, at 6:26 PM, John Lange wrote: I've played around with the lightly documented Asterisk voicemail feature whereby a caller can press "*" during the playback of the OGM and be returned to the "a" extension in the context of the voicemail box. No matter what, Asterisk does nothing when you press "*". It does not interrupt the OGM and it certainly does not return to the "a" context. Watching the console there is no indication of any key press. Has anyone ever got this working? Is there an undocumented setting in asterisk someplace which enables this feature? -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom configuration
You can use one appearance to register which gives you 2 calls with call waiting, then use the others for speed dialing. Or have multiple registrations and use your dial plan to hunt between them. I hear the new 1.0.5 may allow this to happen easier but have not tried yet. On May 13, 2005, at 8:57 PM, Chris Mason wrote: How do you configure your Polycom phones? Is it enough to configure one line appearance? Or is there a way to configure a roll over? Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please share the experience on VoIP phones heavy using.
On Feb 9, 2005, at 9:14 PM, Sergey Kuznetsov wrote: Hi there, Does someone can share the experience with Cisco and Polycom Phones? How rock solid are they? And who will win in sound quality contest? I heard that Cisco phones is a Polycom replicas with changed design. Is that true? What else phones is better to implement to the medium sized business? The rock solid stability and superb sound quality is a must. Both have excellant sound. I think the Polycom speakerphone is a bit better. We are using mostly Polycom these days and our customers love them. My only issue is they do seem to have about a 10% failure rate within 90 days. After that they are solid - so far. They are also less expensive than the Cisco's and seem to have a better feature set and better control of their configs and buttons. I do like the layer 2 troubleshooting capabilities of the Ciscos as the Polycom seem to have no capabilities that I can find. I do not think the Cisco is any kind of a Polycom copy. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 ext. 37 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please share the experience on VoIP phones heavy using.
We buy from both Graybar and Vibes. Not sure but think it took a couple weeks to get phones RMAd. I normally stock my own spares so do not track too close. On Feb 9, 2005, at 10:14 PM, Sergey Kuznetsov wrote: Jerry, Thanks a lot for the feedback! By the way, how long did it take to replace the faulty 10% of phones by RMA? What company did you use to buy it from? Jerry wrote: On Feb 9, 2005, at 9:14 PM, Sergey Kuznetsov wrote: Hi there, Does someone can share the experience with Cisco and Polycom Phones? How rock solid are they? And who will win in sound quality contest? I heard that Cisco phones is a Polycom replicas with changed design. Is that true? What else phones is better to implement to the medium sized business? The rock solid stability and superb sound quality is a must. Both have excellant sound. I think the Polycom speakerphone is a bit better. We are using mostly Polycom these days and our customers love them. My only issue is they do seem to have about a 10% failure rate within 90 days. After that they are solid - so far. They are also less expensive than the Cisco's and seem to have a better feature set and better control of their configs and buttons. I do like the layer 2 troubleshooting capabilities of the Ciscos as the Polycom seem to have no capabilities that I can find. I do not think the Cisco is any kind of a Polycom copy. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 ext. 37 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Q: Does anyone have a WE multi-line card dialer phone working with *?
You have what is a traditional 1A2 type telephone set. If you wish to use the hold key and and lights you will need to connect a 1A2 KSU between it and *. Each line button has the following pr 1 = T/R pr 2 = A lead - this is a closure when the line is offhook pr 3 = Lamp Repeat x 4 Actually the Hold button is probably similarly wired. Hold is accomplished by opening the A lead before the TR loop. This is held in the KSU not the telephone. KSU also supplies your signal battery and lamp battery. Your ringer should be connected to the yellow slate pair of wires. Many other wiring options and configurations but this is the basic. Good Luck On Feb 13, 2005, at 8:40 PM, <[EMAIL PROTECTED]> wrote: Folks, I recently obtained a Western Electric multi-line phone and am seeking help with getting this beast working with *. The interesting stuff in my * implementation consists of a T100P card, a TDM400P card, and an Adtran TA750 channel bank with three quad-port FXS modules and a quad-port FXO. The TA750 is wired to a 24-port Cat 5 patch panel via a 25-pair Amp cable. The phone is a model 2662A1M; it has five lines, a hold button (I presume), card dialer capability, and a 25-pair Amp cable for connecting to The Phone System. (The card dialer feature, IMHO, scores major geek points. If you're not familiar with it, you take a special plastic card about the size of a credit card and punch out two tiny discs for each digit in a phone number. When it's time to call that number, you insert the card in the phone, take the handset off hook, push the "START" button, and--voila!--the phone speed dials your party.) Each line in the phone uses three pairs in the Amp cable; the first pair is for ring and tip, the second pair is a mystery (I'm eagerly awaiting a copy of one of the phone's BSPs so I can find out), and the third pair illuminates the lamp in the button. Most of the remaining pairs in the Amp cable connect to one of the terminal boards inside the phone, and one pair connects to the phone's network (presumably for common ringing, since the leads connect to L1 and L2). If I were to connect the first pair of each line to the patch panel, I would have a perfectly serviceable five-line phone (I haven't yet tried the "hold" button). I would not have, however, illuminated buttons to indicate if channels were in use; nor would the phone ring on an incoming call. If I connect the first and third pairs of a line and plug that mess into a patch panel port, the lamp illuminates and the channel (according to the TA750 and *) goes off-hook--but I do not get a dial tone. I have not, BTW, performed any experiments with a port on the TDM400P. So . . . does anyone have any experience with such a project, or have any ideas on how to trick this up? Cheers, Rob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel DACS and FDL
On Feb 16, 2005, at 3:07 PM, Eric Wieling wrote: I have the following configuration: CLEC -> T-1 -> Asterisk -> Adtran Channel Bank -> (analog) -> Nortel Don't complain that it's ugly. I've already done plenty of that. The CLEC manages their Adtran remotely and needs to be able to continue to do so. I assume they use FDL to do the management. We are using the Zaptel DACS/DACSRBS to cross connect some of the channels directly between the CLEC and the Adtran. These are channels we don't really care about (data, other voice, etc). We are not cross connecting all the channels, just some of them. I'm wondering if/how I can make sure the remote management via FDL continues to work. Does anyone have any information on this or suggestions or anything? What you dropped from your diagram is the T1 from * to the channel bank. FDL is a link level protocol. It is carried in the framing bits of the T1 not within the payload bits. When you use the digium (I presume) card within your * server as a DACS this is connecting the payload, ie timeslot, bits from one port to another. FDL will not work this way. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel DACS and FDL
On Feb 16, 2005, at 7:19 PM, Eric Wieling wrote: Jerry wrote: On Feb 16, 2005, at 3:07 PM, Eric Wieling wrote: I have the following configuration: CLEC -> T-1 -> Asterisk -> Adtran Channel Bank -> (analog) -> Nortel Don't complain that it's ugly. I've already done plenty of that. The CLEC manages their Adtran remotely and needs to be able to continue to do so. I assume they use FDL to do the management. We are using the Zaptel DACS/DACSRBS to cross connect some of the channels directly between the CLEC and the Adtran. These are channels we don't really care about (data, other voice, etc). We are not cross connecting all the channels, just some of them. I'm wondering if/how I can make sure the remote management via FDL continues to work. Does anyone have any information on this or suggestions or anything? What you dropped from your diagram is the T1 from * to the channel bank. FDL is a link level protocol. It is carried in the framing bits of the T1 not within the payload bits. When you use the digium (I presume) card within your * server as a DACS this is connecting the payload, ie timeslot, bits from one port to another. FDL will not work this way. DACSRBS does DACS the robbed bit signalling. I assume this won't help? Ah well. We'll try to find some other way to do Robbed bit refers to robbing a bit from the payload - hence the name - and the reason you can only get a 56k channel on links utilizing this form of signalling - vs 64k when not. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel DACS and FDL
Yes, but does FDL run over the per channel robbed bit signalling or does it run over the T-1 signaling. i.e. Does FDL run using ESF (applies to the whole T-1) or does it run on the robbed bit signaling for specific channels? FDL is carried in some surplus bits within the ESF frame, not within the payload. Follow this link for a good article describing T1/E1s. http://www.commsdesign.com/showArticle.jhtml?articleID=16501900 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZAP FXS vs ethernet FXS
On Feb 21, 2005, at 3:12 PM, Jon Gabrielson wrote: What are the advantages/disadvantages of using a ZAP FXS port versus using one of the many small ethernet FXS devices on the market. The ZAP FXS talks directly to asterisk over PCI. Is this an advantage? The ethernet devices I assume speak either iax2 or sip, does this cripple the functionality of the attached FXS device for things like callwaiting,callerid,distinctive ring, etc... Does anyone have experience with both types of devices and would recommend one over the other? Using a Zap device you are TDM in and out of the card. If you are attaching a FAX or an alarm circuit then this is definately the way to go. Most analog gateways other than the IAXy speak SIP or MGCP. They do still provide your standard features and then some. For best results make certain the ethernet links from the gateways to the server are full duplex and switched, no hubs. Depending on number of ports required, my preference is to use a channel bank, not the cheapest solution always but a highly reliable one. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not answering inbound a line used for outbound
Hi, I've dug a bit through the wiki and the mailing lists, and haven't really seen anything like this, but there must be someone out there doing this. Basically, there is a fax line that I don't want to answer inbound, but I want it available to do dial out from. Right now, we are using a busy wait around the ringing line, but I was hoping for something that might be a little more elegant. Here is the dialplan snip: exten => s,1,Wait(1) exten => s,2,GoTo(s,1) If I'm on the console when a call comes in, it loops through this bit of code a bunch of times. I'm guessing I could lengthen the "Wait(1)" time, but is there any other way to do this? Now, two other related questions: Can Asterisk recognize that something other then a Zap card has a line offhook? (ie there is a fax transmission in progress) Can I do something interesting with CallerID? (for example, create a log of all faxes that called us, or answer certain numbers inside of asterisk) Do excuse me if this is all answered in some concise FAQ, but I've been seriously looking and haven't seen anything about it. Thanks, J ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not answering inbound a line used for outbound
Hi, On Sun Jun 12 02:35:27 CDT 2005 dinesh at alphaque.com wrote: > > On 06/12/05 14:55 jerry at voiptower.com said the following: > > Basically, there is a fax line that I don't want to answer inbound, but I > > want it available to do dial out from. Right now, we are using a busy wait > > around the ringing line, but I was hoping for something that might be a > > little more elegant. Here is the dialplan snip: > > > > exten => s,1,Wait(1) > > exten => s,2,GoTo(s,1) > > that'll still leave the line offhook until the caller hangs up. why not > just use Hangup() to hang up on any incoming call ? I was apparently clear as mud in my first post, sorry about that. This Zap line (FXO) shares the PSTN line with a fax machine. We want to use it as an outbound line for voice calls, but never want to answer it inbound. This is the only bit that is in the start of the context, so the phone is in fact never answered. On the Asterisk console, I get to see it loop around. So, it works. But it just seems crufty to me. Would Hangup() work on an un-Answered() channel? That might be a way to do it, but I think that won't help, as the line will keep ringing. Thanks, J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not answering inbound a line used for outboun
Hi, On Sun Jun 12 09:11:13 CDT 2005, Rich Adamson wrote: > > > exten => s,1,Wait(1) > > exten => s,2,GoTo(s,1) > > > > If I'm on the console when a call comes in, it loops through this bit of > > code a bunch of times. I'm guessing I could lengthen the "Wait(1)" time, > > but is there any other way to do this? > > Sure there is, but without knowing what type of pstn interface your > using, can't suggest an exact coding method. For me (with a TDM-fxo card), > the zapata.conf entry for that channel has a context=nowhere kind of > statement. Since the "nowhere" context doesn't exist, nothing in asterisk > will answer that pstn line. Sorry about the lack of info; I should have mentioned it's an FXO card. Your "nowhere" suggestion seems to fit the bill. Would it be possible to create blank [nowhere] context, so that things are complete, and would that still achieve the same effect? > You can also use one of the fax detect mechanisms (including spandsp) to > sense incoming calls, and if its an incoming fax call send it to a That's an intersting use that I hadn't thought of; but the FXO card is sharing a line with an FXO device, so it shouldn't answer. But, on those lines, I have been reading a bit about Asterisk and Fax, and it's a bit confusing, to say the least. To just answer and recieve a fax, do I need to compile/link against anything else, or will a reasonably current build of Asterisk do this out of the box? (I found a "fax" extension refrence when doing my search for the no answer question, but it is unclear if this is a stub or if all of the functionality is there) > > Now, two other related questions: > > Can Asterisk recognize that something other then a Zap card has a line > > offhook? (ie there is a fax transmission in progress) > > You'll have to clearify the above better. If you're asking about another > bridged phone on the same analog pstn line, no asterisk doesn't currently > have code in it that would recognize speech on the line, lack of a dial > tone, or any other parameter that would suggest the line is in use. (The > TDM card's chipset can detect it, but the code isn't in asterisk to take > advantage of it.) Ok, I'm not sure what you mean by "another bridged phone". I used to have an old-fangled analog phone, and when someone picked up another extension on the same PSTN line, it had an LED that came on, even onhook. So, I'm assuming there is a way to "sense" the line is offhook somewhere. I've read about the dialtone issue, but I think that trying to recognize dialtone/speech is further down the line, once you've gone offhook, but I could be mistaken. > > Can I do something interesting with CallerID? (for example, create a log > > of all faxes that called us, or answer certain numbers inside of asterisk) > > Sure you can. Take a look at the various README files in: > /usr/src/asterisk/doc > and particularily the README.variables file. Well, I checked out Asterisk from CVS, so the path is different, but the files are in there. I briefly looked through this when I installed, but this list and the Wiki have been good refs as of late. Thanks for the reminder though; I had definately forgotten about that dir. To record interesting things through the dialplan though, I'd need some actual context with at least the start extension, correct? (In which case, Could I have it do it's thing and then ignore the incoming call if it doesn't like it?) > However, be careful with CallerID assumptions. Telemarketing and junk fax > generators typically disable sending any CallerID info, so it won't be > all that useful. Yes, I understand that. I'm looking rather at letting paticular people call the fax ine and pick up with asterisk, before the fax machine kicks in. For the accounting stuff, I'm more interested in getting an idea of line utilization (how many incoming faxes, and from where), rather then being able to pinpoint someone sending junkfaxes. Thanks, J ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Questions about contexts
I'm trying to clarify contexts and their uses. I do have a good general understanding of them. My question is about "undeclared" and "non-existant" contexts. If I have a block somewhere (in sip.conf, for example), and it has no "context=thiscontext" field, does it just automatically use the "default" context? Or is this settable? (I see there is an entry for context in the [general] block. I'm guessing this can be used to set a global context for that file. But if this is omitted, what context does the file get? Is "default" a hardcoded fallback?) Someone suggested that I could define a "non-existant" context, being one that isn't actually declared, and set that in my "context=" line, and calls so directed wouldn't go anywhere. Would it be possible to do this with an empty definition (are they equivalent?) Lastly, I'm trying to get a good handle on "include" contexts. If I have: [contexta] ; commands omitted [contextb] ; commands omitted and I add: [contextc] include => contexta include => contextb ; commands omitted Then "contexta" and "contextb" are available to "contextc"'s dialers, right? Do I have to make sure there are no overlapping extensions in all three contexts? Thanks, J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?
> Well voip-info seems to be down. However, a search on google does > bring up info about using Wait(x);... but as I stated.. using that > (even without an answer line) seems to make Asterisk take control of > the line! Apparently you can do this with contexts; I actually _just_ posted a Question for a clarification on this one. (See "Questions about contexts") Basically you aim your inbound Zap channel at either a non-existent context, or a blank context, and Asterisk won't do anything. I'm curious because you say that it "answers" without an Answer line. There are some apps that will automatically answer, but if you are just using wait(x) then there shouldn't be an answer. When you dial into the Zap card, do you hear it answer while watching the console? Thanks, J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXS interfaces
Mike M <[EMAIL PROTECTED]> wrote: > Think "opposite". Green modules are fxs and should be handled with the > fxo signaling. Red modules are fxo and should be handled with fxs > signaling. > > Note the red and green colors here: > http://www.digium.com/index.php?menu=fxsvfxo The opposite thing is hard to catch on to at first, but once you get the hang of it isn't so bad. Plus, the various drivers try to help out with hints if they think the signalling is wrong. > _Don't_ plug a phone into a red module jack. > _Don't_ plug a PSTN line into a green module jack. Silly question: Can you connect an FXO port to an FXS port? Doing this for inter-server exchange would be ugly, but I'm thinking that it might be useful for testing or experimenting. (Maybe not, because it's not simulating a "true" CO interface, but I was wondering if it was possible). As an aside: Why can't you plug a phone into a red module jack? It won't work, but is there anything harmful there? (Not plugging the PSTN into the green jacks is obvious -- the ringing voltage on the PSTN will probably damage the FXS card). Thanks, J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXS interfaces
Hi Alessandro, > But all ports are green! > > p1 -green > > p2 - green > p3 - green > p4 - green I think he means the daughter card color, not the LED on the card slot. What color are the actual daughter cards? J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO as modem (was: * fax reliability between ISDN PRI and FXS ports)
Rich Adamson <[EMAIL PROTECTED]> wrote: > >> I have been investigating on this list and found that faxing is not >> reliable between Zaptel >> cards and that Digium does not support nor >> recommend fax over the TDMXXB interfaces. >> >> Is this true ? Will fax not be reliable enough for my customer ? > > True... the TDM card (or its drivers) has an issue with missed frames > that seriously impacts its ability to handle _any_ modem-type calls. Question: There are problems using FXS ports to pass data and get high speed access, but can an FXO port be used as a modem? Nothing fancy required, 12 or 2400 baud should be ok (9600+ would be nice, but practical over nice). I know there is something like this with the "FAX" extension already. (Not sure if there is a "DATA" equivelent). Is this practical? Has anyone done this? Thanks, J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FXO as modem (was: * fax reliability between ISDN PRI andFXS ports)
Hi Andrew, Andrew Kohlsmith wrote: > > On Monday 27 June 2005 00:34, Jerry wrote: >> Question: There are problems using FXS ports to pass data and get high >> speed access, but can an FXO port be used as a modem? Nothing fancy >> required, 12 or 2400 baud should be ok (9600+ would be nice, but >> practical >> over nice). I know there is something like this with the "FAX" extension >> already. (Not sure if there is a "DATA" equivelent). Is this practical? >> Has anyone done this? > > Sure; this is exactly what rxfax/txfax do. You won't get better than 9600 > baud without running into patent issues, though. Has anyone actually done this, or is it just theoretically possible? I'm not very familiar with the capabilities of the hardware, so excuse me if this is a silly question. I have a basic understanding of how the fax stuff works, but just from reading docs here and there. (I'm not sure how it works under the hood, and indeed I've never played with it yet). I have to ask about the 9600+ patent issues: What are they? I'm just curious. I'm not interested in using the FXO's as a dialin server or anything of that nautre, just to get basic access. Thanks, J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI HDLC Abort (6) Errors
Most T1 circuits are delivered via HDSL2 these days. Hence the single pair. On Mar 4, 2005, at 5:07 PM, Tom wrote: Thanks for the quick reply, we didn't reboot we'll try that, and I've been planning on building a new kernel, I know the fc kernels have issues... I'll report back after I try these two things. On another note we just did some wire tracing and it might be an issue of wiring... We have an adtran card that was installed in our network closet that has a connection back to the NIU in another suite. There is only 1 pair utp going from that adtran card back to the NIU. Anyone have experience with that? We just spoke to our provider and they said there should be 2 pair, but they also said, if there is really only 1 pair, there's no way we'd be able to make calls or keep the connection up... We didn't install the wiring, it was done by the ILEC here, the CLEC who is providing the service said they would send someone out Monday... Anyway, can you run a PRI over just 1 pair with muxing or something? Thanks, Tom Quoting Steven Critchfield <[EMAIL PROTECTED]>: On Fri, 2005-03-04 at 15:27 -0700, Tom wrote: Hello, I have searched and searched, and come up with nothing. I am running Asterisk with a wcte110p configured for t1. Our PRI is staying up, and we can make calls however our service provider's logs are flooding with errors and we are getting lots of HDLC Abort (6) on Primary D-Channel Errors. Our provider says it looks like our box is trying to be the master timer on the circuit (which is not correct they are providing the timing) we have tried both span=1,1,0,esf,b8zs and span=1,0,0,esf,b8zs in zaptel.conf both produce the same problems. The problem is not in "Asterisk" per se as the errors start happening as soon as I modprobe the driver and run ztcfg. As soon as the circuit comes up the errors start on the provider's end. Did you make sure to power cycle afterwords? Sometimes the zap cards don't change critical settings like timing once configured. We are running CVS Asterisk/zaptel/libpri from March 2nd 2005 on Fedora Core 3 fully patched as of last night, I was thinking the problem was with the 2.6 kernel getting preempted and therefore the driver not being able to do its timings right, however fc3's kernels have preemption disabled by default. Does Digium hardware really need/expect a real time OS to run properly? Like I said previously I think the problem is in the driver itself not in asterisk. Any help would be appreciated, and I can code a bit in c so if someone can point me in the right direction I might be able to fix it myself... You probably want to dump the FC kernel like a bad habit. Get a plain vanilla kernel and see if that fixes your problems. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SP300 questions
On Mar 7, 2005, at 5:52 PM, [EMAIL PROTECTED] wrote: Hi, all I have two questions regarding usage of Polycom SP300 with Asterisk. No sure if it is Astersisk or phone related, though. 1. When dialing an extension, one has to perss Dial or Send on the phone after number is entered. Is it possible to avoid this and only enter the number? Configure the digitmap under sip settings to match your dialplan 2. This is probably phone only related, but hopefully someone know the answer. If there wwas a missed call, phone shows "1 call missed". I am trying to figure out how to clear this message from the phone. There are no buttons as far as I can see to get rid of this message on the phone. Goto the Missed Calls directory and exit, the counter will clear. Thanks, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adit 600 for asterisk
On Mar 8, 2005, at 2:14 PM, Steven Critchfield wrote: On Tue, 2005-03-08 at 14:07 -0600, Dennis Webb wrote: Ok, I've pretty much decided to try the Adit route. Somebody who has experience with these tell me if I'm missing something. I have 15 incoming PSTN lines. T1 is not an option at current location. I want to put in an Adit 600 with 2 8-port FXO boards. The adit will then connect to * via a digium t1 board. I configure zaptel.conf for the T1. What other parts would be needed? How do the PSTN lines connect to the Adit, standard rj11 jacks? It looks to be about a $2500 investment and I need to know if there is anything special I am missing. The Adit has a 50 pin D connector that is like the old SCSI connectors. To connect your phone lines, you either want to get a punch down block that has a 50 pin connector on it, or possibly a device with RJ11 style jacks and a 50 pin connector. Then you connect it with a 25 pair cable with 50 pin connectors on either side. Go to any reputable supplier near you and they should be able to help you look at and find what you are comfortable with to use for installation. -- Install an RJ21 and cable to your Adit. As Steve mentioned the Adit has 2 50pin connectors. They are female. An RJ21 is a 66M150 Block mounted on a bracket and wired to 2 50 pin connectors mounted on the same bracket. These connectors may be either male or female, but generally female. See http://www.siemon.com/e-catalogXML/ datasheet.aspPN=157C&FAM=S66PrewiredMSerie&P=5739 for an example. Go to your local telecom distributor, probably who you are ordering your Adit from, and get a prewired block and 2 25pr cables to connect your adit. If you are not going to ever exceed 24 lines then you may only connect 1 25pr able. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 Phantom Ringing
Never had any of my 100 or so act like that. What version of code are you running? I think 1.4.1 is the latest. On Mar 8, 2005, at 8:05 PM, Ben Ruset wrote: Hello list: I have a very odd problem. Seemingly randomly, my Polycom IP600 phones will ring without a call being placed to it. That is to say, a random phone will ring. Nothing shows up under Caller ID. Even the buttons that light up to show an incoming call do not light up. If you pick up the handset, you can hear the phone ring through the speaker. Hanging up the phone makes it stop ringing. Then, sometime later, it will happen on another random extension. Is this a common problem? Where can I look to start diagnosing this? Thanks! -ben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAC Access Bank Manual
Carrier Access generally have all of their manuals available for download. You just have to request a free login. they also provide excellent dialin support - also free. If your framing LED is blinking I would double check that both ends of your span are set for ESF. zttool is the tool for working on the cards. On Mar 17, 2005, at 4:40 AM, Vicky Shrestha wrote: Hi, Does anyone have Carrier Access Corporation (CAC) Access Bank I Manual ? Could you please email it to me off list ? We have a FXS channel bank and the framing Error Led is blinking and I have no clue on what could be the problem . Is there command line utilities available in Linux to Troubleshoot T1 connection using Zaptel drivers ? /etc/zaptel.conf = span=1,1,0,esf,b8zs #span=1,1,0,esf,ami #span=1,1,0,d4,b8zs #span=1,1,0,d4,ami #e&m=1-24 fxols=1-24 loadzone=us defaultzone=us == /etc/asterisk/zapata.conf = [channels] language=us context=default signalling=fxo_ls ;usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes ;threewaycalling=yes transfer=yes ;cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel => 1-24 === dmesg output = Zapata Telephony Interface Registered on major 196 Found TE410P at base address dfcdff80, remapped to d0e23f80 TE410P version c01a009b, burst ON FALC version: 0005, Board ID: 00 Reg 0: 0x0e3c6800 Reg 1: 0x0e3c6000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1000 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx Registered tone zone 0 (United States / North America) TE410P: Span 1 configured for ESF/B8ZS SPAN 1: Primary Sync Source == -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom vs. Cisco IP Phones
The lack of full SIP suport and the cost of Ciscos license plus the added base cost of their phones moved us away from Cisco and over to Polycom. They have been working extremely well. Software updates are free and the update process is relatively simple. I have found the IP600 is a great desk phone. I do use some IP300s for break rooms etc, but do not like them on a desk. Not sure the rquirements for your receptionist. I have found that the IP600 does have most everything required to function properly. If you do have an office without DID and a lot of traffic then you may want to look at the tools to display status on her computer. I do have a Snom inhouse for testing when I get a chance. Their quality however is not as good as either Cisco or Polycom. On Mar 17, 2005, at 11:03 AM, Max Clark wrote: Hi all, I am working on building a new VoIP PBX. Looking at the current market for phones it seems my best "enterprise" options are the Cisco and Polycom phones. I have some experiance with the Cisco 7940G, but the process of flashing the phone with the SIP firmware left a bad taste in my mouth (not to mention the added expense for the phone). What is the general consensis about the polycom IP phones? Are they good? Are they better than Cisco? What do I do for the receptionist's station? Thanks in advance, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Include/Macro not working right...
I think that should be
#INCLUDE numbers.conf
Note the caps and no ""
On Mar 17, 2005, at 12:31 PM, [EMAIL PROTECTED] wrote:
Hey guys. Thanks for the help on the Pattern matching, I got that
working pretty nicely.
the next problem I have is that I'm using an include file, but its not
really working...
In my extensions.conf:
[incoming]
exten => _NXXNXX,1,SetCallerID("Unknown Called Number")
#include "numbers.conf"
exten => _NXXNXX,3,Macro(Number,1000,${EXTEN})
[macro-Brand]
exten => 1000,1,Playback(${EXTEN})
And my numbers.conf:
exten => _5551234567,2,SetCallerID("Bob's Seafood")
exten => _5557891234,2,SetCallerID("Damon Stern's Garage")
When I reload, I can see my include load right, but...
When I call 5551234567, I get the SetCalledID("Unknown Called Number")
then I get the Auto Fallthrough, status in 'UNKNOWN' and it hangs up.
I'm sure I'm doind something stupid. Please correct me. ;-) Tahnks
in
advance.
Steve
Stephen Amadei
5114 Harbor Beach Blvd
Brigantine Beach, NJ 08203
(609) 703-9649
Current resume at http://www.amadei.com/resume.doc
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Re: [Asterisk-Users] What cable to connect TE110P to telco PRI ?
Straight through patch cord works well. Assuming you are going from the card to the telco NIU. Otherwise you may need to make a crossover cable. Pins 1,2,4,5 are the relevant ones. Most BRI these days are single pr so no. On Mar 17, 2005, at 4:55 PM, Robert Rozman wrote: Hi, call me stupid, but cable is not delivered with te110p. What cable can I use to connect to telco NT PRI line ? Is it same as for BRI interface ? Thanks, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAC Access Bank Manual
On Mar 18, 2005, at 2:40 AM, George Pajari wrote: Vicky Shrestha wrote: The asterisk configuration and the channel bank configuration are both set to esf and b8zs. Howerver I am still getting the framing Error "Red and blinking". zttool shows there are no alarms. According to the manual, Framing Error (Red and Blinking )means "Network T1 is out of frame (received signal cannot be framed to ESF or D5 as configured by T1 Option switch 4)" I tried with both DIP switch on and off, but no help. Any ideas ? Is my card or channel bank bad ? Probably not. More likely your CAC ABI is set up to run in TR08 mode which is incompatible with standard T1 framing. Check the LIU board (the Line Interface Board -- as opposed to the FXO or FXS cards) and look at the PROM. It usually will be marked TR08. If that is the case you will need to order a D4/ESF upgrade kit. If you have a 1.x revision TR08 chip, you will need P/N 750-0018. If you have a 3.x revision TR08 chip, you will need P/N 750-0019. Are you using a terminal to talk to the CAC? Depending on configuration you may have the DIP switches disabled and changing them will do no good. Connect to the CAC and ask it for the T1 configuration to verify what it really is. Also make sure you rerun ztcfg after any changes to zaptel.conf. What does zttool tell you? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Firmware
I would suggest contacting a dealer until you find one who will sell you a maintenance contract for the phone. Last I checked, over a year ago, they were somewhere around $10-$20. Once you have a contract you may register online and download all the software you need. However to use legally you would have the pay the ~$50 license fee for SIP. On Mar 18, 2005, at 10:17 PM, Patrick M. Gray, Jr. wrote: That seems to be what the various documents I've stumbled across seem to indicate. Maybe it's too late at night and my brain is shot, but Cisco's documentation on the upgrade path seems a little confusing... Would you mind giving me a brief summary of the upgrade path to the latest firmware if you know it, starting from P003AM30? Thanks again! Pat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Friday, 18 March, 2005 22:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP Firmware Just a note that you will need to perform quite a few incremental upgrades to get to a current firmware version. So if you do get someone who will sell you the firmware, make sure you get the all of them. On Fri, 18 Mar 2005 22:04:29 -0500, Patrick M. Gray, Jr. <[EMAIL PROTECTED]> wrote: I got a new old stock Cisco 7960 from eBay and the warranty expired bay in 2001 according to Cisco (I didn't even know they had VoIP in 2001 ;-) ). I spoke with a wonderfully rude gentleman at Cisco who told me there was nothing that could be done to get SIP firmware for the device, and would not even entertain the possibility of purchasing said FW from Cisco. He suggested I call a local reseller, and the single one I called was not interested in helping me either with my "unsupported hardware." I'm using the 7960 to experiment with *, and was wondering if there are alternative means to finding the firmware, or if the "out of the box" SCCP firmware (I have version P003AM30) will work with *. I'm willing to pay any "official resellers" a fair price for the F/W, but the attitude I received from Cisco and the one reseller I contacted have me thinking this is a waste of time. I'm using [EMAIL PROTECTED] and can seem to find any SCCP info, and don't want to delve too deeply into this experiment if the phone is not going to work reliably. Thanks for any help or pointers in the right direction. Pat ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context
On Mar 26, 2005, at 9:38 PM, AS wrote: How do I how to send a call as [EMAIL PROTECTED] ? Not exactly sure what you are asking. If you are trying to dial a specific extension within a specific context then I use a GoTo. [currentcontext] exten => 8885551212,1,GoTo(anothercontext,100,1) [anothercontext] exten => 100,1,Dial(whatever) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rhino Channel Bank or ADIT 600
I don't know what you mean by talk. Cost difference: well this is how I calculated it, one the overall cost of having to get an asterisk box that needs more CPU, and 2 the actual cost of the cards to make it work. As well as taking into considiration that adit is telco grade equipment, and has very good performance. Option 1: An adit 600 loaded with 6 FXS cards = $800, Digium Quad t1 = $1500. Option 2: An adit 600 with 5 FXS cards, and 1 CMG02 card = $1500 (Max, you could get it for cheaper). Option 1 for 384 (8 * 48) analog ports: 8 * 800 + 2 * 1500 = $9,400. (around $24.50 per port) Option 2 for 360 (9 * 40) analog ports: 9 * 1500 = $13,500. ($37.50 per port) But option 2 can all go on one asterisk box, since it's all voip, and no transcoding has to take place (in fact it's like getting 9 more asterisk boxes to give you some more horsepower). With option 1 you will run into problems of using more than one digium card on one box, although it works, it is not recommended. Consider that a dual xeon system costs $3000 you will end up paying $7,700 for each pair of 4 adit boxes, and this doesn't even give you the options of adding telco t1s. In my opinion using the CMG cards will pay out in every single way when you are talking of anything more than 2 Adit boxes to a single system. Couple observations. Adit with CMG uses MGCP vs SIP. Not sure how extensive the * support is for this. I believe Carrier Access is working on a SIP release but not sure how complete it will be. CMG/CMG2 are nice cards. Also have a nice license builtin for G729. If you use though you will need a matching one for the * server. However there is also a call limit of 12, I think, for the CMG and 24 for the CMG2 cards. Depending on application you could easily exceed this. So take the 8 T1 capacity with a grain of salt, you will not be able to have that may calls up at a time. Also one of my favorite applications is connecting an Adit directly via T1 to a Digium card. Then using another T1 port on the Digium card to connect to PRi from PSTN. This is called a traditional TDM switch. No IP in the patch, no headaches. Great for such things as FAX and ALARM circuits which are very problematic with IP in the path. It is not always about up front costs but about capabilities and support costs down the line. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Push VLAN to Polycom via DHCP
On Mar 27, 2005, at 12:10 AM, Matt Darnell wrote: Has anyone been succesful pushing a VLAN setting to a Polycom phone via DHCP? Chicken or the egg! How can the Polycom reach the proper DHCP server if it is not on the correct VLAN? That's why Ciscos and Polycoms support CDP, so the CDP-capable switch can supply the correct voice VLAN. I 'assumed' the phone would reboot with the new VLAN setting and get a new IP address from the DHCP server on the phone VLAN - there would be two DHCP servers. I can't think of any other way to make it work with DHCP. If it isn't designed to work that way, why would they put the option in the DHCP section. -Matt I had always understood that they only supported VLAN discovery via CDP. But reading the 1.4 admin guide it says this... VLAN ID See 2.2.1.2.2 DHCP Menu on page 7 Special Case: Cisco Discovery Protocol (CDP)a overrides Local FLASH which overrides DHCP VLAN Discovery. a. Can be obtained from a connected Ethernet switch if the switch supports CDP. This seems to imply that DHCP can be used to spec a VLAN. I too would like to find a way to make this work. I also do not think the phone would need to reboot. I have noticed being able to change the VLAN and have the tag applied or not without the phone rebooting. Actually about the only thing I can do and not have it reboot:-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SP300 questions
Just Look at the missed calls screen and exit, the counter will clear. Not sure on an IP300, but on an IP600 fastest is to press down arrow then left arrow. On Mar 28, 2005, at 6:32 PM, Paul Hales wrote: 1. You can set up items in the Digitmap (under SIP conf) to know when a number is complete. 2. The up and down arrows let you look through the missed calls. Later, PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 8 March 2005 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom SP300 questions Hi, all I have two questions regarding usage of Polycom SP300 with Asterisk. No sure if it is Astersisk or phone related, though. 1. When dialing an extension, one has to perss Dial or Send on the phone after number is entered. Is it possible to avoid this and only enter the number? 2. This is probably phone only related, but hopefully someone know the answer. If there wwas a missed call, phone shows "1 call missed". I am trying to figure out how to clear this message from the phone. There are no buttons as far as I can see to get rid of this message on the phone. Thanks, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] constant ringing on Zap channels
Since signalling info is carried in the A/B bits which is how * talks to the Adit regarding the state of each channel, any framing misconfig or timing misconfig will cause this. Perform a print config on the adit and closely compare with zapata.conf and zaptel.conf On Mar 29, 2005, at 8:02 AM, Andrew Kohlsmith wrote: On March 29, 2005 08:40 am, Richard Reina wrote: This goes on continuously and no phones are ringing. I am using a digium T1 card and ADIT 600. Do you have the Adit600 configured correctly? It's not stuck in a test mode or anything? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help w/ basics
On Mar 29, 2005, at 4:58 PM, Garrett Nelson wrote: Hello, I am new to Asterisk and new to this list. I got Asterisk setup and running using [EMAIL PROTECTED], and purchased a PolyCom SoundPoint IP500 phone to test out. I cannot get the phone to talk to the Asterisk box. On bootup of the phone, it tells me that it cannot contact boot server. Why is that? It gets an IP fine, and I have also tried manually setting the IP of the phone and the Asterisk system in the phone’s menu. Per your list below you do not have a boot server. You also do not need one. Read the admin guide for the phone to learn how to configure it. I don’t know what I am doing, and don’t know where to look for help. I have tried the wiki but didn’t find anything and just got more overwhelmed. Can anyone point me to an easy guide to getting my phone to talk to my Asterisk system, or help me set things up? What have you actually setup in your sip.conf? I do not use @home but hear it is very easy to setup. Network configuration: 192.168.1.1 IPCop 1.4 firewall/router (DHCP server) 192.168.1.2 Windows 2003 server 192.168.1.20 Asterisk system 192.168.1.200 PolyCom SoundPoint IP500 phone ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help w/ basics
On Mar 29, 2005, at 5:34 PM, Rod Bacon wrote: Unfortunately, the polycom phone was a bad choice from a beginner's perspective, as it loads it's firmware/config from a boot server every time it's powered up, and won't work without it. Get a softphone (firefly, xten) running on a PC first (to ensure that your asterisk is OK). Take your Polycom back to the shop, and get a cheap-ass Grandstream Budgetone. Uhhh - which Polycoms are you talking about? All the ones I have work great without a boot server. Just manually enter network info and let it boot. then point your web browser at it and set your config info. All info is saved in internal memory and upon reboot will behave as before. A boot server is great if you have many telephones to support, but for just a few don't worry about one. Except you will need to have either an FTP or tftp available to upgrade the firmware or the bootrom code. Yes a Grandstream is an OK beginners phone. In fact it is the one I started expeimenting with. However since I was investigating using for enterprise class applications I soon found it was not a good fit. Enjoy your Polycom!!! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 Cannot answer
I don't think you want both dynamic and defaultip set
But that should not cause what you describe. I hvae seen other issues
with head. Perhaps checkout the latest?
On Mar 30, 2005, at 12:29 AM, MDS wrote:
I googled and googled but could not find anything regarding this
problem.
I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6
months, no problems with my grandstreams. I'm fairly familiar with the
ins and outs of asterisk...
IP600 with latest sip 1.4.1 and bootrom from my FTP server.
Standard config files from http://www.freedomphones.net/polycom/files/
No changes other than typical ip address of phone and server.
Grandstream (192.168.2.20) is exten 2000, Polycom (192.168.2.22) is
2006.
I can make calls out to my Grandstreams from the Polycom all day. No
problem.
When I try to call the Polycom I get this stuff:
-- Executing Dial("SIP/2000-972f", "SIP/2006|10|r") in new stack
-- Called 2006
-- SIP/2006-f8ea is ringing
-- SIP/2006-f8ea answered SIP/2000-972f
-- Attempting native bridge of SIP/2000-972f and SIP/2006-f8ea
-- Got SIP response 481 "No Such Call" back from 192.168.2.20
== Spawn extension (from-sip, 2006, 1) exited non-zero on
'SIP/2000-972f'
-- Got SIP response 500 "Internal Server Error" back from
192.168.2.22
-- Got SIP response 500 "Internal Server Error" back from
192.168.2.22
When I answer the polycom it just hangs up and hangs the grandstream
online. I have to manually hang up the grandstream. It doesn't get a
SIP
notifcation of call failure or hangup.
When I tcpdump the asterisk box, I can see RTP streams from the
Grandstream toward the server. But nothing coming from or toward the
Polycom. When I call the Grandstream from the Polycom, the call
connects
and I see both RTP streams to and from the Asterisk box for both phones
and everything is happy.
anyone have any ideas as to why inbound calls fail?
I've tried several combinations of
friend/peer/progressinband/canreinvite etc... No change at all.
Here's my sip.conf for the Polycom
[2006]
type=friend
username=2006
secret=2006
host=dynamic
dtmfmode=rfc2833
defaultip=192.168.2.22
progressinband=no
context=from-sip
[EMAIL PROTECTED]
callgroup=1
pickupgroup=1
thank you for any insight!
Mark
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Re: [Asterisk-Users] What is ZAP ? newbie question sorry
ZAP are channels which connect through hardware boards installed within the * server. If only using softphones feel free to not use;-) On Mar 30, 2005, at 7:32 AM, iMRAN wrote: Hi Pros, Please advice whats the purpose of ZAP, if i have softphones and ATA 186 with PSTN trunk, wht ZAP will do ? do i zap to route calls internal softphone to softphones ? thnx a lot Ronny ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] username/password for PolyCom IP500 webinterface?
Check the admin guide but if memory serves simultaneously pressing 4 6 8 * will wipe out to factory default. May have to be done while booting. On Mar 30, 2005, at 11:31 AM, Wiley Siler wrote: It is... Polycom 456 The setup for using new confs and app files is done through the phone anyway. Just setup the FTP server and your files. Then at least you should be able to get the latest app file son the phone to ensure it works right, even if not configured correctly. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garrett Nelson Sent: Wednesday, March 30, 2005 9:38 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] username/password for PolyCom IP500 webinterface? try this sir,Polycom SpIp --- - - Tried that, didn't work. Is my phone just messed up? Is there way I can change that password through the phone itself? Is there a way to reset the phone to factory settings? I know how to reboot it but didn't see a way to reset everything. -Garrett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
On Mar 31, 2005, at 8:01 AM, Zoa wrote: cpu load on te4xxp cards is very low, and now that they have echo cancellers as add-ons cards, it will be even lower. I can't speak on hardware compatibility as i never tried a sangoma card. (But i can say that in the last year i've never had an issue with digium cards and we have 8 in use.) The te405p card resolved most incompatibilty issues. Digium has a hardware echo can? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap tuning for echo/gain
I'm having a bit of an issue with one of the bargain x100p clones, and I'm not sure what the right approach is. My symptom started as way loud offset delayed echo from voip hardphones -> PSTN through the clone card. I played with and then learned everything I could about echo cancelling, and have managed to bring the echo down to acceptible levels, where it's not too intrusive. In my travels I discovered "ztmonitor", and thought I'd run this. Now, with no call and no activity on the line, the RX side shows a reading of about 1/3 of the bar graph. I figured this is bad (right?) and tweaked my rxgain value (I had never had to touch this before) until ztmonitor only showed one bar. I accomplished this by adding a negative value to the rxgain, I believe it was -12. My question is: Is this the right approach? Should I be tweaking my gains first, and then adjusting echo, or vice versa? Is there something else completely I should be looking at first, or is this the right thing to do? I did have a genuine Digium analog TDM card in there previously, but I had to swap boxes and for some reason the TDM isn't probing (and digium support haven't been too helpful, but that is another story). The TDM didn't need any tweaks; it seemed to have worked out of the box (although I might have been more tolerant of slight echo before, but the clone card was simply intolerable for conversation). Any suggestions or comments welcome. Thanks, J. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip registration question
Hi everyone, I have a number of SIP registrations going fine, but am trying to get a new provider going, and they have no sample Asterisk SIP config. They have been helpful, but keep falling back to the way they "think" packets should be flowing, and I've been trying to figure out how the Asterisk config should look like to get the SIP packet to look correct. Now, they say that from a phone this works fine, and that our config must be at issue. The claim is that "Asterisk isn't doing MD5 authentication right", and since I'm not an expert with SIP MD5 auth in asterisk, may be true. Right now, I'm trying to get the registration happening. On a "test" server, we've been able to put through a call w/o registration, so it seems some of this can be compatible. I'm wondering if I can use "md5secret" with a "register => " statement. The current busted config: [general] ;register => userid:pass:[EMAIL PROTECTED]:5069 [myipsolution] type=friend authuser=acctid username=userid secret=pass md5secret=XXXMD5HASH of userid:asterisk:pass X nat=yes host=voipprovider.com port=5069 insecure=very canreinvite=no The error on the console is: Jul 16 11:29:20 NOTICE[3361]:-- Registration for '[EMAIL PROTECTED]' timed out, trying again Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong password on authentication for REGISTER for 'userid' to 'voipprovider.com' The password is right, as given and verified by the provider. Any suggestions would be great. Thanks, J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip registration question
Hi, Quoting Michiel van Baak <[EMAIL PROTECTED]>: > On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote: > > > > The error on the console is: > > Jul 16 11:29:20 NOTICE[3361]:-- Registration for '[EMAIL PROTECTED]' > > timed out, trying again > > Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong password on authentication > > for REGISTER for 'userid' to 'voipprovider.com' > > Did you try to put the md5 encoded password in your > register=> line ? I didn't before (I wasn't sure that was a valid syntax) ... but I have tried now, same error. Is there something to tell asterisk to try an MD5 auth, either in the password or on the registration line? Thanks for your quick response. J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap channel(s), meetme and codecs/licences
Hi all, Some simple questions about codecs: What codec does the Zap channel use by default? Can this default be changed, and to what? (g729 too?) What codec does meetme use? (I think this is ulaw, but asking to be sure) Can you use another codec, or does everything have to be transcoded to ulaw? Finally ... if I have a 3way call going, between 1 g729 caller and two other callers, do I need one or two available licences? (I'm guessing that zap doesn't do g729, and am wondering if I have an FXO caller and a local FXS person talking to a VoIP caller using g729, how it would work) Trying to nail down how all of the codecs and figure out how the licence costs might work. Thanks, J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap channel(s), meetme and codecs/licences
Hi Rich, Quoting Rich Adamson <[EMAIL PROTECTED]>: > > What codec does the Zap channel use by default? > > None/all. Look in /usr/src/asterisk/configs/zapata.conf.sample and > you won't find any reference to codecs. I did better; I googled, searched the wiki, and the list archives. I find this hint from "zap show channel #": Default law: ulaw I didn't know if this was just a documentation thing, or if it was something that was configurable. > > Finally ... if I have a 3way call going, between 1 g729 caller and two > > other callers, do I need one or two available licences? (I'm guessing that > > zap doesn't do g729, and am wondering if I have an FXO caller and a local > > FXS person talking to a VoIP caller using g729, how it would work) > > Someone else might want to chime in here, but it seems to me (as a non- > programmer) that internal handling of voice packets (within *) were > primarily slinear or something like that. If two end points can communicate > with the same codec, the voice data is simply passed through (no > conversion). If one user is g729 only communicating with other users, > that g729 user consumes one license instance to convert to whatever > the other users might be using. Two g729 users and one g711 user, likely > uses two g729 licenses. This is my thinking too, but that is what I'm a bit unclear about. > Without a better understanding of what devices you are truly trying to > use, I don't believe anyone is going to be able to answer your questions > relative to zap channels and codec selection/conversion. I have a customer that wants to try using a local IAX provider for origination. The provider is using g729, and right now I have some digium FXS ports (along with an FXO for outbound access). We have some SIP hardphones, and they all have the ability to do g729. We will need a licence to do voicemail unless we convert files to g729 format. However, to get a phonecall to be handled by the Zap phones, I wasn't sure if something was needed or not, and you've cleared that up. I asked about meetme internal format just so I could see if there was a possibility to "work around" the need for converters. My worry is that we might run out of licences during the test, and that Asterisk would simply drop a call if it can't complete one leg of the call, leaving both the caller and my customer frustrated. Ideally we could buy n+1 licences, but I am trying to get some sort of a feel for what "n" will look like, and how practical/viable that is on a much larger scale rollout. Thanks, by the way... you are always very helpful, both in the several replies to my messages, and many other messages replies to others. J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap channel configuration problem
Hi Alexis, Quoting "Alexis F." <[EMAIL PROTECTED]>: > I would like to use a digum card to call an external number through my > PSTN. I think that I have a problem in the configuration. Asterisk > returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' > > I make modprobe zaptel && modprobe wctdm without error return. Does it report finding the card? > Configuration files : > > /etc/zaptel.conf > fxsks=1 > loadzone = it > defaultzone=it > > > /etc/asterisk/zapata.conf > [channels] > language=en > context=from-sip > signalling=fxs_ks I don't see a channel definition line there. I think it should be something like: channel => 1 but depends on your hardware config. Asterisk definately doesn't know about your zap, and that is why you are getting that error. To be sure, issue a "zap show channels" in the CLI. J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR disposition field always says ANSWERED on inbound calls
Hi,
Quoting Giorgio Incantalupo <[EMAIL PROTECTED]>:
> My inbound context is:
>
> [inbound_menu]
> include => internals ; very strange: include doesn't work!!!
> exten => _X.,1,DigitTimeout(2)
> exten => _X.,2,Answer
> exten => _X.,3,NoOp(DS:${DIALSTATUS}) ; my debug purpose
> exten => _X.,4,Background(fga_main_menu)
> exten => _X.,5,Background(3-sec-pause)
> exten => _X.,6,Background(fga_main_menu)
> exten => _X.,7,Hangup
>
> exten => 101,1,Macro(interni,${PIPPO},${RING_TIME})
>
> The Macro executes a Dial commandso nothing strange.
> >>anybody knows why the CDR field named "disposition" always says ANSWERED
> >>on inbound calls even if nobody picks up the phone (we are using various
I do believe this might clarify; if not, you are explicitly answering
the line. I don't see any Dial commands (might be in your include),
but if your call hits this extension it will be considered answered
after priority 2 (if I understand how the CDR works). You are then
having them go through an IVR, but the call is already deemed connected.
I think you want to play with NoCDR or one of it's cousins to get the
effect you desire.
J.
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RE: [Asterisk-Users] Snom 200
David: I talked to one of the key people at ABP a few minutes ago. They had a snow storm in Dallas today, and everything is closed down. If you send an e-mail to [EMAIL PROTECTED], they are working from home, and will respond. They will be in the office tomorrow. Thanks, Jerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Davis Sent: Wednesday, February 26, 2003 6:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Snom 200 Does anyone know who sells the Snom 200 (or a better suggestion) other than ABP(http://www.abpintl.com/)? They aren't answering their phones and email today and I need to get 5 of these sent out very soon. David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intercom and Paging
James: The Snom phones have an auto answer mode setting that answers in the speaker phone mode. However, once it's set, it automaticly answers every call. Also, the audio on the speaker phone mode in the Snom 200 is much, much, much better than the Snom 100. Pingtel also has a java app you can load into their phone which allows it to do the same thing, but only for a pre-programmed (in the phone) set of numbers. If the number is not on the list, it just rings normally. Hope that helps, Jerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James H. Thompson Sent: Thursday, February 27, 2003 6:07 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Intercom and Paging Anyway to get paging/intercom functions using asterisk with SIP or ADSI phones? On my current phone, I can intercom another extension (or all extensions) and make an announcement, the person at the other end can talk back without touching their phone (if its set to handsfree intercom answer). Can any of the SIP or ADSI based phones do this? Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ip phones and intercom/paging
Title: Message I may be entering this thread a little late, but just in case I'm not, the Snom 100 and 200 both support auto-answer very nicely. Jerry G. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cwitteSent: Friday, August 08, 2003 9:36 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] ip phones and intercom/paging There was a thread a few months ago that tossed around some ideas for using a cisco phone for intercom or paging. I don't have any ip phones, and wondered if anyone had any luck getting intercom or paging to work on the cisco units. Do any of the (cheaper) ip phones have a way to support intercom or paging? I presume that it's not part of the SIP or IAX protocols. Chris. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] exiting Voicemai for VoiceMailMainl
Title: Message Has anyone been able to exit voicemail2 to get to voicemailmain2 in order to check voicemail messages from a remote phone? I can dial a "0" while my voicemail intro plays, and * dumps the call and says it was sent into invalid extension "o". I set up extension "0", and I can call it from any phone on the system. I can transfer any phone to it. The "0" extensionis set up to enter voicemailmain2. It works fine. Is there another way to exit during the voice mail prompt in order to check your voic mail? Thanks, Jerry
RE: [Asterisk-Users] Overhead Paging
Title: Message We do the same thing with the Snom phones. They can be set up for auto-answer, and they have a speaker jack in the back that is the same levels as a sound card on a PC. And the Snom phone automaticly hangs up when the caller hang up is detected (the SIP BYE message). Jerry -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805)Sent: Thursday, November 13, 2003 6:17 PMTo: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Overhead Paging Our setup is to set the OSS device to autoanswer. The output of the soundcard feeds into a bank of overhead speakers. If the channel is in use, then the call gets put in a queue until the OSS device is free. -sb -Original Message-From: Johnson, Randy [mailto:[EMAIL PROTECTED]Sent: Thursday, November 13, 2003 5:34 PMTo: '[EMAIL PROTECTED]'Subject: [Asterisk-Users] Overhead Paging Does anyone have any recommendations for overhead paging systems for use with Asterisk? Thanks, Randy Johnson
RE: [Asterisk-Users] Overhead Paging
Title: Message Scott: The operation you describe with multiple phones set to ring on the same extension is correct. The first one that answers, gets it. However, the meetme setup you describe also works great. I have that set up for a conference bridge where one person sets up a conference with one click on a web page which calls multiple Snom phones into a conference. This is a full conference where everyone can talk to everyone. However, The Snom phone also allows you set it up with the mic permanently muted, which would work great for paging. Jerry -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805)Sent: Friday, November 14, 2003 9:25 AMTo: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Overhead Paging Jerry, Do you have it setup so that multiple phones answer one extension? I tried that setup with two Cisco phones, however, only the quickest responding phone answered. If you have a config that rings multiple phones and all of the phones answer the same call, I'd be interested to see the config. I guess theway to do it would be to setup a meetme conference and then dial all parties into the conference then speak -sb -Original Message-From: Jerry Gibson [mailto:[EMAIL PROTECTED]Sent: Friday, November 14, 2003 8:52 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Overhead Paging We do the same thing with the Snom phones. They can be set up for auto-answer, and they have a speaker jack in the back that is the same levels as a sound card on a PC. And the Snom phone automaticly hangs up when the caller hang up is detected (the SIP BYE message). Jerry -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805)Sent: Thursday, November 13, 2003 6:17 PMTo: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Overhead Paging Our setup is to set the OSS device to autoanswer. The output of the soundcard feeds into a bank of overhead speakers. If the channel is in use, then the call gets put in a queue until the OSS device is free. -sb -Original Message-From: Johnson, Randy [mailto:[EMAIL PROTECTED]Sent: Thursday, November 13, 2003 5:34 PMTo: '[EMAIL PROTECTED]'Subject: [Asterisk-Users] Overhead Paging Does anyone have any recommendations for overhead paging systems for use with Asterisk? Thanks, Randy Johnson
RE: [Asterisk-Users] Overhead Paging
Multicast would be great if all of the SIP clients would accept it. Of course, first you have to do call setup via SIP. I hvaen't ween that done yet for multiple phones for the same call. Jerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Albertson Sent: Friday, November 14, 2003 11:54 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Overhead Paging I'd hate to see conference bridging use for paging. A lot of wasted CPU and bandwidth. Could you "multicast" the UDP packets? We assume you don't need to page across multiple Asterisk servers but if you did the software wuld need to be smart enough to "know" which groups of extensions could be in a multicast and whci need to be bridged. Basically check to see if the SIP phone are on the same subnet. --- DUSTIN WILDES <[EMAIL PROTECTED]> wrote: > I feel this needs to be a separate application in Asterisk, like > app_sipintercom The application would connect to all available > auto-answer SIP phones, play a short frequency tone for the intercom > alert, only allow one-way streaming to the phones, then disconnect all > phones whenever the originator hangs up. > > Same is true for a paging application, app_sippage > The application should work the same as intercom, but allow two-way > audio streaming. > > I was starting the design of these two applications unless anyone else > has a better idea or has already begun work? Feedback welcome > > > > > > -Original Message- > From: Jerry Gibson [mailto:[EMAIL PROTECTED] > Sent: Friday, November 14, 2003 10:41 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Overhead Paging > > > Scott: > > The operation you describe with multiple phones set to ring on the > same extension is correct. The first one that answers, gets it. > However, the meetme setup you describe also works great. I have that > set up for a conference bridge where one person sets up a conference > with one click on a web page which calls multiple Snom phones into a > conference. This is a full conference where everyone can talk to > everyone. However, The Snom phone also allows you set it up with the > mic permanently muted, which would work great for paging. > > Jerry > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, > Scott (7805) > Sent: Friday, November 14, 2003 9:25 AM > To: '[EMAIL PROTECTED]' > Subject: RE: [Asterisk-Users] Overhead Paging > > > Jerry, > > Do you have it setup so that multiple phones answer one extension? I > tried that setup with two Cisco phones, however, only the quickest > responding phone answered. If you have a config that rings multiple > phones and all of the phones answer the same call, I'd be interested > to see the config. I guess theway to do it would be to setup a meetme > conference and then dial all parties into the conference then > speak > > -sb > > -Original Message- > From: Jerry Gibson [mailto:[EMAIL PROTECTED] > Sent: Friday, November 14, 2003 8:52 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Overhead Paging > > > We do the same thing with the Snom phones. They can be set up for > auto-answer, and they have a speaker jack in the back that is the same > levels as a sound card on a PC. And the Snom phone automaticly hangs > up when the caller hang up is detected (the SIP BYE message). > > Jerry > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, > Scott (7805) > Sent: Thursday, November 13, 2003 6:17 PM > To: '[EMAIL PROTECTED]' > Subject: RE: [Asterisk-Users] Overhead Paging > > > Our setup is to set the OSS device to autoanswer. The output of the > soundcard feeds into a bank of overhead speakers. If the channel is > in use, then the call gets put in a queue until the OSS device is > free. > > -sb > > > > > -Original Message- > From: Johnson, Randy [mailto:[EMAIL PROTECTED] > Sent: Thursday, November 13, 2003 5:34 PM > To: '[EMAIL PROTECTED]' > Subject: [Asterisk-Users] Overhead Paging > > > > Does anyone have any recommendations for overhead paging systems for > use with Asterisk? > > Thanks, > Randy Johnson > > = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Solved! Snom 200 Busy signal
-Original Message- >From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Lawson >Sent: Thursday, November 20, 2003 7:03 PM >To: [EMAIL PROTECTED] >Subject: [Asterisk-Users] Solved! Snom 200 Busy signal >As a follow up to my earlier posting, the problem with the Snom 200 Busy >signal was the firmware! I reverted back to 1.16x and everything's OK. >That made today pretty complicated, since I already had a new kernel and >a new Asterisk build I was trying all at once >Hopefully someone else will benefit... >- Matt I had the same type of problem you did when I first upgraded to 2.02t. But I did a CVS update on 11/19/03, and everything is fine now. What I did notice (and I have a customer that saw the same thing) is that that version 2.02t has added a new SIP "Publish" message. I still haven't been able to determine what they're publishing with it. But Asterisk didn't seem to like it until I did a CVS update. It didn't seem to respond to it. Now Asterisk sends back a "Status: 405 Method Not Allowed" message, and everything is fine. I can't definitively say this was the cause because I didn't go back to the old Asterisk code and verify. But I can tell you that my Snom 200 phones with version 2.02t work fine now. Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk AGI and DTMF
All, I have my AGI working. I am placing a call in the outgoing directory and running my AGI. Once the call is places and answered I then need to send DTMF tones. Like 101. How can I do this in the AGI? I did not see and commands for it. Thanks, Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI startup on channel when asterisk starts
All, I am looking for a way to have my AGI startup on a channel automatically when asterisk starts. Is this possible? I have my AGI working for when a call comes in - however I would like the AGI started up automatically with asterisk on a couple channels as I want to monitor my database and when things happen place a couple calls etc. I am aware of the outgoing directory but that is not exactly what I am wanting to do unless there is a way to have the commands in the file not actually dial but just give me a channel and start my AGI. They my AGI can place the call if that is what is required. Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] adding additional volume to console/dsp
Is there a setting that can add additional volume to the Console/Dsp output? 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
[asterisk-users] conf meetme all exited and still active
Running asterisk 1.2.23 on a meetme two polycom SIP phones. When we both hung up the meetme was still active. Is there something special needed to destroy the static conference? 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
[asterisk-users] play greeting from odbc voicemail
I have voicemail configured to store messaging in an odbc database. Does anyone have any thoughts on how best to play back someones greeting from the db? Thanks, ~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
[asterisk-users] ata device but for a soundcard
I am looking for an ATA like device but instead of VOIP to analog phone I want VOIP to low level audio out. Something that looks like a sound card output. I know I can use cheap PC's but that then you have HD's to setup etc... HD failures etc... Anyone know of something like that? 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
[asterisk-users] question on call setup
If I have an asterisk server and lets say 150 asterisk clients all running SIP and I wish to initiate a one way call out and bring those 150 units into a one way conference/ intercom group/ something ... What is the fastest method? how long would that call setup take before I can start talking? Is there anything else I'm not thinking of? Thanks, 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
[asterisk-users] Page() command
Will the Page() command handle 200 SIP devices? How much time does it take for ALL 200 devices to be ready to receive audio? 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
[asterisk-users] console dsp
I am trying to get a console/dsp application going with 1.4.18 and not hearing any audio. In the CLI I see the call coming in, I see the Dial(Console/dsp) I see I see ALSA default but I hear no audio. What can I do to tell what is happening here. I have in modules.conf: noload chan_oss.so load chan_alsa.so For kicks I tried it the other way to noload chan_alsa.so and load chan_oss.so and same result no audio. THanks, 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] console dsp
Jerry Geis wrote: > I am trying to get a console/dsp application going with > 1.4.18 and not hearing any audio. In the CLI I see the call coming in, > I see the Dial(Console/dsp) > I see > I see ALSA default > > but I hear no audio. > > What can I do to tell what is happening here. > > I have in modules.conf: > noload chan_oss.so > load chan_alsa.so > > For kicks I tried it the other way to noload chan_alsa.so and load > chan_oss.so and same result no audio. > > THanks, > > Jerry > More information: I can play wav files just fine on the soundcard. its an nforce motherboard. I am using centos 5 x86_64. 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
[asterisk-users] no audio between two asterisk servers
HI I am running asterisk 1.4.18 on both machines. I have two asterisk machines setup and a polycom phone, the main machine is #1 the polycom phone is connected to it as the server. I come offhook on the polycom, dial 21, which calls machine #2. Machine #2 is only to play demo-congrats. Watching the asterisk monitor on both machines I see on #1 the Dial going to machine #2, machine #2 shows executing playback(demo-congrats). However, I hear no audio. I have done "service iptables stop" on both machines. Both machines have in their sip.conf entry disallow=all allow=ulaw allow=alaw Everything "appears" to be working - just not hearing audio. Why else would I not hear audio? When I run the polycom and call into the dialplay on machine #1 I hear audio just fine. 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
[asterisk-users] zaptel compile question
Hi all, I am wanting to use an option from the ./configure script with zaptel to compile zaptel for a different kernel than the running kernel. How do I do that exactly. Example: Current kernel is 2.6.18-8.1.4.el5 and I want to compile zaptel for 2.6.18-53.1.4.el5 I am using centos 5.1 or RHEL 5.1 So when I reboot zaptel is ready to go. How do I do that? Thanks, 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
[asterisk-users] setting callerid across servers
I have a situation when a T1/PRI line comes into box 1 then uses SIP over to box 2 and all my phones are on box 2. if the person is not at their desk on ring no answer I am calling their cell phone which places the call back over SIP to box 1 and out the T1 . How can I setup this configuration so the original caller ID will show up on the cell phone. Thanks, 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] 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
[asterisk-users] using the System() command to call a script
I am wanting to call a shell script in System() command and it does not seem to work. If I put a direct linux command it does work. My script is owned by root and is executable. Is this possible? What might I be missing? 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
[asterisk-users] MeetMe option b
I am running asterisk 1.4.18 trying to use MeetMe and option b. I am getting permissions denied failed to execute conf-background.agi on the CLI lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi -> /home/silentm/bin/conf-background.agi my conf background is a symbolic link - then my permissions are : [EMAIL PROTECTED] src]# ls -l /home/silentm/bin/conf-background.agi -rwSr-Sr-- 1 root root 81 Mar 17 10:44 /home/silentm/bin/conf-background.agi I have tried with just 744 and also with 744 and chmod +s. Any ideas why I would get the permission denied? 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] MeetMe option b
Jerry Geis wrote: > I am running asterisk 1.4.18 trying to use MeetMe and option b. > > I am getting permissions denied failed to execute conf-background.agi > on the CLI > > lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi -> > /home/silentm/bin/conf-background.agi > my conf background is a symbolic link - then my permissions are : > > [EMAIL PROTECTED] src]# ls -l /home/silentm/bin/conf-background.agi > -rwSr-Sr-- 1 root root 81 Mar 17 10:44 > /home/silentm/bin/conf-background.agi > > I have tried with just 744 and also with 744 and chmod +s. > > Any ideas why I would get the permission denied? > > Jerry > I seem to have gotten past the permissions error by putting #!/bin/sh # at the top of my script file... Now when I run permissions issue is gone, however, When I try to access by variables (as I have dont many other times) I am getting an error. In my call file I have SetVar: MEETME_PLAYFILE=/home/silentm/record/pc.610.wav And when I asking for the value of MEETME_PLAYFILE it is giving an error about the PIPE being broken. Is there something special about this option "b" and what an AGI can and cannot do??? The AGI actually calls a C program (this is the same C file I have been using for a long time) and I just ask the AGI for the values of the variables. 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
[asterisk-users] MeetMe option b
I have been playing with meetme option b. I was wanting to use a spool call file going to a local channel, then bring a list of asterisk boxes into a meetme in talk only mode. then run the AGI to play a wave file then hangup. It seems to be doing all this but I hear no audio. I seem to get pipe errors when running the AGI. Should this work? Is there an example of this anywhere. I basically just want to bring in a list of asterisk boxes (console ports on their end) speak a wave file and hang up the meetme. seems easy enough. This same thing works for speaking live. All I added was playing a wave file instead with the agi. 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
[asterisk-users] Way to hangup a call between two asterisk machines
I am using a call file with a channel like: Channel: SIP/machine1_to_machine2/1042 for example. This is between two asterisk boxes. When this is done machine1 does not show anything for "core show channels". however machine 2 does show the channel as being active. How can I keep the call showing on machine 1 in "core show channels" so when I need to kill or hangup the call I can. Eveything is working in my testing, I just need to call to show up on machine 1 so I can hangup the call when I need to... probably through an agi. 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
[asterisk-users] question how to play wave file in meetme and exit
Hi all I am playing with meetme and I have things working in speak live mode. What I would like to do is have a small meetme that kicks off on a schedule (no problem there) and just plays a wave file in the meetme and then exists. I have played with option b of the meetme where the AGI plays a wave file. However it seems like I have not audio. I see things happening as expected on the CLI screen but I have not audio. Should this work? The meetme is pure SIP. I do have a zaptel TDm800 in my server but this particular meetme is pure SIP. Is there an example of this somewhere? the agi is being called, the SIP devices are in the meetme I just dont hear audio. The AGI reports is played the file "200" return code but I actually returns really fast like it did not play the wav file. I can play the given wave file in asterisk for other uses. If I cannot use meetme for this how else would I do it? Bring X SIP devices in a meetme and play a wave file then exit. Thanks, 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
[asterisk-users] question on meetme
I am trying to use meetme() on SIP channels. I found this line on voip-info.org - It *is* necessary either to have a Digium card or a dummy timing driver (e.g. ztdummy or zaprtc) in order for MeetMe to work at all, but that doesn't help you use AGI with SIP channels: They have no capacity to use any AGI script at all. If they try to, they get no audio. - I am using 1.4.18 and I am hearing no audio. just like it says. I do have a digium TDm800 card in my box. Does asterisk 1.6beta have this same issue still How do people get around this issue? I am looking for a meetme conferencing in a handful of phones, call AGI to play a wave file then hangup My meetme is a pure SIP meetme. Thanks, 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
[asterisk-users] Chanspy from a call file
How would I setup chanspy from a call file??? I want to make a call file that uses a Local channel and plays a wave file. Then I want to make a call file that calls an extension and does a chanspy on the local channel above that is playing a wave file. How do I set that up? 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
[asterisk-users] question on app_conference()
MeetMe() has the K option that kills the conference, how do I do that in app_conference() as there no kill the conference option? 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] 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" ;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=63872&sid=aff61bbd5ddeea61bc831239b220db23
>
>
>
> Anyone have any bright ideas on what might be wrong and/or
> troubleshooting tips?
>
>
>
> …brig
>
> --
>
> Please direct emails to [EMAIL PROTECTED] or call
> 816-767-5549. This will help with issues getting full exposure to
> the dept and allow for the quickest response.
>
>
>
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asterisk-users
[asterisk-users] Help with cisco 7960 phone
I have a cisco 7960 phone. Worked fine in the office. I took it home. At home I have a linksys router that the phone is plugged into. The linksys router has DHCP enabled. I am getting the following error on the console from the 7960. I have tried it with nat=yes and nat=no in the sip.conf file. --- Transmitting (NAT) to 192.168.1.69:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.69:5060;branch=z9hG4bK1415360297;received=192.168.1.69;rport=5060 From: "Display Name" ;tag=1683635072 To: "Display Name" ;tag=as4c59a734 Call-ID: [EMAIL PROTECTED] CSeq: 3091 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a1c350c" Content-Length: 0 -- The username and secret are the same as they were in the office when it worked. I figure it has to be something easy but I have not found it yet. the sip.conf entry for this phone is: [570] type=friend dtmfmode=rfc2833 username=570 secret=XXX disallow=all allow=ulaw allow=alaw host=dynamic context=local-sip callerid="Home 570" <570> nat=no What might I try to get the phone working from home? Thanks, 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
[asterisk-users] help with no audio
I am using asterisk 1.4.18 with a polycom phone.
sip.conf has:
[532]
type=friend
username=532
secret=XXX
dtmfmode=RFC2833
host=dynamic
context=smvoice-sip
callerid=532
qualify=no
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
canreinvite=no
I call into the dialplan and try to play demo-congrats and I hear nothing.
Firewall is disabled.
Everything is on the 192.168.1.X network for this simple configuration.
The tftp server is giving the polycom phone the config files.
Any ideas why I dont hear audio?
Jerry
---
Use 'exit' when done
Asterisk 1.4.18, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <[EMAIL PROTECTED]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
== Parsing '/etc/asterisk/asterisk.conf': Found
[0;37;40m[1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found
[0mConnected to Asterisk 1.4.18 currently running on demobox (pid = 18129)
demobox*CLI>
Verbosity is at least 5
[Kdemobox*CLI>
-- Executing [EMAIL PROTECTED]:1] Playback("SIP/522-051fc8f0",
"demo-congrats") in new stack
?-- Playing 'demo-congrats' (language 'en')
?
[Kdemobox*CLI>
== Spawn extension (smvoice-sip, 10, 1) exited non-zero on 'SIP/522-051fc8f0'
?
[Kdemobox*CLI> sip set debug
demobox*CLI>
SIP Debugging enabled
[Kdemobox*CLI>
<--- SIP read from 192.168.1.99:5060 --->
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bK725666cbFE41ACBC
From: "522" ;tag=87113650-18E1B969
To:
CSeq: 1 INVITE
Call-ID: [EMAIL PROTECTED]
Contact:
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249
v=0
o=- 1207070053 1207070053 IN IP4 192.168.1.99
s=Polycom IP Phone
c=IN IP4 192.168.1.99
t=0 0
m=audio 2228 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<->
?--- (14 headers 11 lines) ---
?
[Kdemobox*CLI>
Sending to 192.168.1.99 : 5060 (no NAT)
?Using INVITE request as basis request - [EMAIL PROTECTED]
?
[Kdemobox*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.99:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.1.99;branch=z9hG4bK725666cbFE41ACBC;received=192.168.1.99
From: "522" ;tag=87113650-18E1B969
To: ;tag=as47ea6357
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5ac2b96f"
Content-Length: 0
<>
?
[Kdemobox*CLI>
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method:
INVITE)
?Found user '522'
?
[Kdemobox*CLI>
<--- SIP read from 192.168.1.99:5060 --->
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bK725666cbFE41ACBC
From: "522" ;tag=87113650-18E1B969
To: ;tag=as47ea6357
CSeq: 1 ACK
Call-ID: [EMAIL PROTECTED]
Contact:
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Max-Forwards: 70
Content-Length: 0
<->
?--- (11 headers 0 lines) ---
?
[Kdemobox*CLI>
<--- SIP read from 192.168.1.99:5060 --->
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bKb6f79a4aC708EF33
From: "522" ;tag=87113650-18E1B969
To:
CSeq: 2 INVITE
Call-ID: [EMAIL PROTECTED]
Contact:
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="522", realm="asterisk", nonce="5ac2b96f",
uri="sip:[EMAIL PROTECTED];user=phone",
response="9a7bd42e9bbf18fef41b63bccc83178c", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249
v=0
o=- 1207070053 1207070053 IN IP4 192.168.1.99
s=Polycom IP Phone
c=IN IP4 192.168.1.99
t=0 0
m=audio 2228 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 t
Re: [asterisk-users] help with no audio
> > On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote: > >/ I call into the dialplan and try to play demo-congrats and I hear nothing. > />/ > />/ Firewall is disabled. > />/ Everything is on the 192.168.1.X network for this simple configuration. > />/ The tftp server is giving the polycom phone the config files. > />/ > />/ Any ideas why I dont hear audio? > / > Do you happen to have an unconfigured T1 card in your machine? That's > the most common problem I see for people when they get no audio at all > coming out of Asterisk. If that's not the case, I'd turn RTP debugging > on in the command-line and make sure RTP packets are coming and going > from the Asterisk box. > > -- > Jared Smith > Community Relations Manager > Digium, Inc. Jared, I have no card in this unit at this time. lsmod shows ztdummy loaded. I turned on "rtp debug" and got a bunch of lines like: Got RTP packet from 192.168.1.99:2226 Does that help you? 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
[asterisk-users] call files
I am trying to use call files that dial and play a wave file on 3 asterisk boxes console dsp. This is working. The 3 boxes are noticeably out of sync. From using 3 different call files (time to process) I'm sure is the time delay. Is there a way to get these audios more in sync? 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] help with no audio
Jerry Geis wrote: >> >> On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote: >> >/ I call into the dialplan and try to play demo-congrats and I hear >> nothing. >> />/ />/ Firewall is disabled. />/ Everything is on the 192.168.1.X >> network for this simple configuration. >> />/ The tftp server is giving the polycom phone the config files. >> />/ />/ Any ideas why I dont hear audio? >> / >> Do you happen to have an unconfigured T1 card in your machine? That's >> the most common problem I see for people when they get no audio at all >> coming out of Asterisk. If that's not the case, I'd turn RTP debugging >> on in the command-line and make sure RTP packets are coming and going >> from the Asterisk box. >> >> -- >> Jared Smith >> Community Relations Manager >> Digium, Inc. > > Jared, > > I have no card in this unit at this time. > lsmod shows ztdummy loaded. > > I turned on "rtp debug" and got a bunch of lines like: > Got RTP packet from 192.168.1.99:2226 > > Does that help you? > > Jerry > Jared, Using the "rtp debug" I noticed that when the phone has no audio all I see is: Got RTP packet ... Got RTP packet... There are no Sent RTP packets.. under normal cases there is One sent and one Got: Got RTP packet... Sent RTP packet... Why would asterisk not be sending RTP packets I have no hardware card in this test system. Just two polycom IP330 phones. Once in a great while I will hear audio when calling into the dialplan and playing demo-congrats. 95% of the time I hear NO audio though. I am using asterisk 1.4.18, libpri 1.4.3 and zaptel 1.4.9.2 (ztdummy is loaded) Why might asterisk NOT be sending RTP packets? 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
[asterisk-users] RTP no sound on asterisk
Hi all, I seem to only be getting (1) call to sip_write() in channels/chan_sip.c I have a very simple setup. one server (no cards) 2 polycom IP 330 phones. Server is 192.168.1.150 and phone is DHCP. Nothing else on the network. No firewall is enabled. I call into the dialplan with: exten => 112,1,Answer exten => 112,n,Playback(demo-congrats) exten => 112,n,Hangup I see this executing on the CLI. However I have no audio. Enabling RTP debug I see the Got RTP packet but there are no send RTP packets going out. I edited the source and put logging messages first in main/rtp.c and I saw the ast_rtp_raw_write() getting called 1 time. so I backed up the tree. Got into channels/chan_sip.c sip_write() and it only gets called 1 time. I have had a couple of times where I heard audio. Hangup up and tried again. And NO audio for bunch more times... What can be causing my RTP issue and no audio? 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] help with no audio
Jerry Geis wrote: >> >> On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote: >> >/ I call into the dialplan and try to play demo-congrats and I hear >> nothing. >> />/ />/ Firewall is disabled. />/ Everything is on the 192.168.1.X >> network for this simple configuration. >> />/ The tftp server is giving the polycom phone the config files. >> />/ />/ Any ideas why I dont hear audio? >> / >> Do you happen to have an unconfigured T1 card in your machine? That's >> the most common problem I see for people when they get no audio at all >> coming out of Asterisk. If that's not the case, I'd turn RTP debugging >> on in the command-line and make sure RTP packets are coming and going >> from the Asterisk box. >> >> -- >> Jared Smith >> Community Relations Manager >> Digium, Inc. > > Jared, > > I have no card in this unit at this time. > lsmod shows ztdummy loaded. > > I turned on "rtp debug" and got a bunch of lines like: > Got RTP packet from 192.168.1.99:2226 > > Does that help you? > > Jerry > I have found the echo command. I modified the dialplan to use echo. I turned on rtp debug and I see packets going BOTH ways. I have looked all through the zaptel.conf (below) everything is commented out. there are no cards in my box. zapata looks the same everything commented out. I am not finding a reason for not getting audio packets sent back to the phone. Any suggestion on something to try? Jerry - # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a sync source, just use "0" # # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) # # The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1 # # Note: "d4" could be referred to as "sf" or "superframe" # # The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1 # # E1's may have the additional keyword "crc4" to enable CRC4 checking # # If the keyword "yellow" follows, yellow alarm is transmitted when no # channels are open. # #span=1,0,0,esf,b8zs #span=2,1,0,esf,b8zs #span=3,0,0,ccs,hdb3,crc4 # # Next come the dynamic span definitions, in the form: # dynamic=,,, # # Where is the name of the driver (e.g. eth), is the # driver specific address (like a MAC for eth), is the number # of channels, and is a timing priority, like for a normal span. # use "0" to not use this as a timing source, or prioritize them as # primary, secondard, etc. Note that you MUST have a REAL zaptel device # if you are not using external timing. # # dynamic=eth,eth0/00:02:b3:35:43:9c,24,0 # # Next come the definitions for using the channels. The format is: # = # # Valid devices are: # # "e&m" : Channel(s) are signalled using E&M signalling (specific # implementation, such as Immediate, Wink, or Feature Group D # are handled by the userspace library). # "fxsls" : Channel(s) are signalled using FXS Loopstart protocol. # "fxsgs" : Channel(s) are signalled using FXS Groundstart protocol. # "fxsks" : Channel(s) are signalled using FXS Koolstart protocol. # "fxols" : Channel(s) are signalled using FXO Loopstart protocol. # "fxogs" : Channel(s) are signalled using FXO Groundstart protocol. # "fxoks" : Channel(s) are signalled using FXO Koolstart protocol. # "sf" : Channel(s) are signalled using in-band single freq tone. # Syntax as follows: #channel# => sf:, # rxfreq is rx tone freq in hz, rxbw is rx notch (and decode) # bandwith in hz (typically 10.0), rxflag is either 'normal' or # 'inverted', txfreq is tx tone freq in hz, txlevel is tx tone # level in dbm, txflag is either 'normal' or 'inverted'. Set # rxfreq or txfreq to 0.0 if that tone is not desired. # "unused" : No signalling is performed, each channel in the list remains idle # "clear" : Channel(s) are bundled into a single span. No convers
Re: [asterisk-users] help with no audio
On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote: >/ I have no card in this unit at this time. />/ lsmod shows ztdummy loaded. / Just to make sure that this is not the problem, what's the output of: zttest -c 3 -- When running this nothing comes back... It says "Opened pseduo zap interface, measuring accuracy..." and that is all. I am using Centos 2.6.18-53.1.14.el5 I also just tried rmmod ztdummy and then starting asterisk again and the audio works. something is wrong with ztdummy. I went back to zaptel-1.4.9.2 diretory and did make clean; ./configure; make; make install (one at a time ) I saw no errors. tail /var/log/messages after modprove showed no errors. Now what? 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
