Re: [Asterisk-Users] Help Required for Speex
On Sat, 2003-11-22 at 17:01, God Knows Well wrote: Hi Have any body experienced Asterisk with Speex??May i know the result i.e Voice quality or echo problems and wats frame size and other settings are compataible with asterisk . Before you rub someone the wrong way, please understand that no one here is Required to answer any of your questions. You may request help and you may seek assistance, but please do not come to this forum demanding support. Also, you should understand that there is a great wealth of information on this subject in the archives. The archives have been indexed regularly by google for some time now. You really should consult this resource first so your question can be directed to get specific information. Also you might wish to consult the source code. It isn't difficult to read even from a non programmers point of view. At least try and use grep over the source to find certain pieces. You should find out from the list that echo is not a function of codec, but rather interfaces and devices. Frame size is probably 20ms. Settings seem to be settable via the source if you wish. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Asterisk - Nikotel disconnects after 1 Minute
Hello list!
I'm using Asterisk CVS-11/22/03-04:28:51 and try to route my normal
(classic) phone calls via nikotel (www.nikotel.com). I can talk about 1
minute and get then disconnected. Here my current configuration parts
which affect nikotel:
register = chabrol:[EMAIL PROTECTED]/500
[nikotel]
type=friend
secret=PASSWORD_REMOVED
username=chabrol
fromuser=chabrol
host=calamar0.nikotel.com
qualify=1000
context=internal
I also tried the register without /500 because there are no calls routed
inwards via nikotel and configured the type type=peer. Additionally I
tried to set auth=md5 and left off the qualify parameter. But it
changed nothing.
In the extension file i use:
[chabrol]
include = internal
exten = _00N.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Any ideas?
Best regards,
Daniel
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Re: [Asterisk-Users] Newbie ... some questions
I seriously doubt these things are possible.. not without recoding some of the Asterisk components.. The whisper thing might take some work, but wouldn't it be possible to forcibly park a call and have the manager pick it up to achieve #2? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI problems
It seems that there's a non-printable character at the beginning of the DNIS stream I'm getting from the SUMA 4 switch. Once I chopped that off, everything works right. Would you mind sharing with the rest of the list your patch to drop off the character? Or was it simply $EXTEN:1? How did you identify the nonprintable? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P Errors under load
For anyone trying to make an E1/T1 crossover, here's a nice diagram from NMS that may help. The only pins that are needed in a short cable for testing are 1,2,4,5. Aren't those the only wires that need to be connected for _any length_ of cable since the others are non-connects on the jacks? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callerid problem...zaptel ppl
i am having callerid problems with *. i have the callerid from my telco and it shows up in my normal phone when i connect it directly to the line but if i connect the same phone thru * server the callerid is not shown. i am using X101p and tdm400p. i have everything defined in my zapata.conf well and fine. i finally came to conclusion that this might have happened due to the registration of country with zaptel in the zaptel.conf. i am in nepal. if anyone could help, that would be great. I have a friend with two phone lines (Bell Canada) -- both with CID, both plugged in to their own X101P card. One line gets CID just fine, the other has problems (1 in 6 calls have CID come through for *). A regular old phone in either picks up CID every time. Swap the lines so that they're going into the other X101P; problem stays with line. Change cabling up to demarc, no change in problem. The only thing I can come up with is that the FSK softmodem in * is a little too strict, and the one line is a little too loose. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk suitable for this use?
Do you really want all those spans going down cause someone tripped over a power cable or your hard drive nukes itself? You usually don't worry about either of those problems when you've got redundant power supplies and drives in the rackmount system in a locked room. We only use 2 TE410Ps in our systems and many servers. This way you spread out the load and achieve redundancy at the same time. What do you use for servers? What's the load like? I wasn't aware that PCI could handle 8 full PRIs of traffic. What codecs are you using? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk suitable for this use?
How's this worse than an as5300? I could install ata-flash and get high-ish end pc hardware (rcc serverworks boards, etc). Heck, if it was such an evil thing, I'm sure Voicepulse (along with every other clec) and friends wouldn't be doing it! The dialup ISP I worked at used MaxTNTs with DS3s... 665 ports would die if someone tripped over both power cords or someone zapped the DS3 cables. Mind you the whole system was backplane and hotswap... I guess a bigger question is how many TE410P cards can PCI handle? I mean my rough figuring would say this: DS1 = 8kB/sec * 24 = 192kB/sec PCI = 33Mhz * 4 bytes/clock = 132MB/sec I get 132MB/sec / 192kB/sec or 687 DS1s __IF__ you're not spewing that data back out the PCI bus. Let's say that there's no codec translation going on and whatever comes in is going out again... so halve that or 343 DS1s, again strictly theoretical. From discussing on the IRC channel this morning the TE410P can bus master one frame from all spans at once; Burst transfers are PCI's bread and butter so if you have bus-mastering network and storage devices too, I imagine you'd run into CPU issues from codec translation before you'd run into PCI bandwidth issues... Anyone please feel free to jump in and correct me. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk suitable for this use?
On Sunday 23 November 2003 09:38, Harry McGregor wrote: On Sun, 2003-11-23 at 07:57, Andrew Kohlsmith wrote: Do you really want all those spans going down cause someone tripped over a power cable or your hard drive nukes itself? You usually don't worry about either of those problems when you've got redundant power supplies and drives in the rackmount system in a locked room. And redundant UPSs as well, two good sized UPSs can really help. As far as cases go, I like the Bow cases (www.gogobow.com, you can buy some of them from www.newegg.com). We only use 2 TE410Ps in our systems and many servers. This way you spread out the load and achieve redundancy at the same time. What do you use for servers? What's the load like? I wasn't aware that PCI could handle 8 full PRIs of traffic. What codecs are you using? Sure, 8 PRIs are only 8x 1.5megabit x2 (ie all of it going out your network connection), that is 24 megabit. 33MHz 32Bit PCI can handle 132 MegaBytes/sec, so the PRIs (doubled) only account for a small amount of the bandwidth available. Standard 33mhz/32it PCI can handle routing DS3 and OC3 level bandwidths. I would recommend 33mhz or 66mhz 64bit PCI if you are going above that. Unless I'm mistaken, there's another barrier you're going to hit before that: the zaptel drivers have a maximum of 252 channels addressable (each channel gets its own device minor number and 4 device minors are reserved in the driver for other purposes). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download
Third and last question for now: The phonebook used to be in the ini-file, but it seems to be somewhere else now ??? I'd like to preprogram other entries in there :-) It's a separate file now. I have put in a feature request to have TWO phonebooks paths in the ini file, a global and personal (with personal overriding global for duplicate entries) so that you could handle a global directory and still be able to have personal entries without a LOT of duplication work. And if you could hit these files from HTTP it's be absolutely wild. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callerid problem...zaptel ppl
--- Andrew Kohlsmith [EMAIL PROTECTED] wrote: i am having callerid problems with *. i have the callerid from my telco and it shows up in my normal phone when i connect it directly to the line but if i connect the same phone thru * server the callerid is not shown. i am using X101p and tdm400p. i have everything defined in my zapata.conf well and fine. i finally came to conclusion that this might have happened due to the registration of country with zaptel in the zaptel.conf. i am in nepal. if anyone could help, that would be great. I have a friend with two phone lines (Bell Canada) -- both with CID, both plugged in to their own X101P card. One line gets CID just fine, the other has problems (1 in 6 calls have CID come through for *). A regular old phone in either picks up CID every time. i can't get callerid in anyway thru *. Swap the lines so that they're going into the other X101P; problem stays with line. Change cabling up to demarc, no change in problem. whats demarc? can u explain more? The only thing I can come up with is that the FSK softmodem in * is a little too strict, and the one line is a little too loose. :-) with throrough readings and more, i found that there is a caller id program which could be used to check if my modem supports the callerid format coming from the telco. my zaptel modem doesnot seems to detected by linux as a modem. how can i find in which port my modem(X101P) is installed? /dev/ttys0 once this is achieved i can tweak some portions in callerid.c to work with our countrys format. the format i guess is similar to UK...callerid comes before the first ring??? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download
Hi, - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, November 23, 2003 6:22 PM Subject: Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download Hi Dan, Some preliminary testing with this version makes me very happy :-) Great to hear that :-) However, is there an option so that the sound-dialog can be forced to just use the defaults it preselected for me ? At first run, you are asked to select the audio devices... one time only.. Then it will be used at each run Secondly, can you share the algorithm for the username/password/server codes, so I can build some pre-install packages that can just be installed/unzipped and run 'out of the box' for the user ? Yup, I'll send it to you by mail directly. It is just used for a regular user not to drop an eye in the file, not for cryptography experts..:-)) I can incorporate something more elaborated in the future, if neccessary.. Third and last question for now: The phonebook used to be in the ini-file, but it seems to be somewhere else now ??? I'd like to preprogram other entries in there :-) There is a separate file now, named diax.pb, using the same format as the previous one. This is because the config file will be updated automatically with any new version starting with 0.9.5 You will be able to replace just the executable and the help file and keep all previous configured cfg, cl and pb files. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download
Hi, - the phonebook is now in a separate file; Duh, I can't read :-) :-)) Another issue I've just seen, however :-) When I'm passing a call from a Zap channel (PRI) I get an error: STATUS: Bad or incomplete voice This is strange someone else with this issue? I've also noticed this with iaxComm, so I don't think it is DIAX specific, but I would _love_ to know what it's about :-(( This is library related (both clients use the same Steve's iaxclient library) Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download
Hi, I have put in a feature request to have TWO phonebooks paths in the ini file, a global and personal (with personal overriding global for duplicate entries) so that you could handle a global directory and still be able to have personal entries without a LOT of duplication work. I intend in 0.9.5 to be able to have a secondary phonebook file (global) into another directory or even a share which will be merged with the local one And if you could hit these files from HTTP it's be absolutely wild. :-) Is possible. Keep on eye on the development ;-) Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download
Citeren Dan [EMAIL PROTECTED]: However, is there an option so that the sound-dialog can be forced to just use the defaults it preselected for me ? At first run, you are asked to select the audio devices... one time only.. Then it will be used at each run Yes, this was clear to me. However the defaults are usually fine - and if they are not, the person will most likely know about it because he has to manually configure anything that uses soundcards :-)) I'd be fine with it just accepting defaults if something was found to be out of order.. I'd be happy to know if others have different views on this though... Third and last question for now: The phonebook used to be in the ini-file, but it seems to be somewhere else now ??? I'd like to preprogram other entries in there :-) There is a separate file now, named diax.pb, using the same format as the previous one. This is because the config file will be updated automatically with any new version starting with 0.9.5 You will be able to replace just the executable and the help file and keep all previous configured cfg, cl and pb files. Cool. Makes sense. -- Met vriendelijke groet, Florian Overkamp ObSimRef BV ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callerid problem...zaptel ppl
On Sun, 2003-11-23 at 08:52, Andrew Kohlsmith wrote: i am having callerid problems with *. i have the callerid from my telco and it shows up in my normal phone when i connect it directly to the line but if i connect the same phone thru * server the callerid is not shown. i am using X101p and tdm400p. i have everything defined in my zapata.conf well and fine. i finally came to conclusion that this might have happened due to the registration of country with zaptel in the zaptel.conf. i am in nepal. if anyone could help, that would be great. I have a friend with two phone lines (Bell Canada) -- both with CID, both plugged in to their own X101P card. One line gets CID just fine, the other has problems (1 in 6 calls have CID come through for *). A regular old phone in either picks up CID every time. Swap the lines so that they're going into the other X101P; problem stays with line. Change cabling up to demarc, no change in problem. The only thing I can come up with is that the FSK softmodem in * is a little too strict, and the one line is a little too loose. :-) If the problem stays with the line, have you noticed any other differences with the line? Just guessing here, hopefully this will give you ideas to move forward on. There is a zapmonitor app that comes with the zapata library, it dumps the raw audio from the line to the soundcard. Try using it to verify if the callerid spill happens where the library is expecting it too every time. I think on my mothers Bellsouth line her callerid sometimes doesn't get sent till into the 3rd ring. This would possibly cause problems as I think asterisk is set to start extension processing after the 2nd ring. So if this is the problem, you may be able to alter asterisk to wait for callerid spill. Hope this helps. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callerid problem...zaptel ppl
i can't get callerid in anyway thru *. If * can't get it how do you think you're phone's gonna get it if it's conneected via *? whats demarc? can u explain more? point of demarcation -- where the telco says anything up to this exact spot, we'll look after and take responsibility for. Anything after is your problem. Usually a box in the basement or on the outside wall of your house. The first place I look for trouble related to lines is to disconnect everything on my side of the demarc and hook the * box up there directly. If it still don't work then you've eliminated a LOT of head-scratching. with throrough readings and more, i found that there is a caller id program which could be used to check if my modem supports the callerid format coming from the telco. my zaptel modem doesnot seems to detected by linux as a modem. how can i find in which port my modem(X101P) is installed? /dev/ttys0 once this is achieved i can tweak some portions in callerid.c to work with our countrys format. It's a glorified softmodem. You don't do that. Start up * with -gc and look at the console when a call comes in. It should come between the first and second rings. (I'm speaking of North America here). the format i guess is similar to UK...callerid comes before the first ring??? UK CID is not detected by * at this time, which is what all that free-running FSK modem process talk was about. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download
Hi, - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, November 23, 2003 7:39 PM Subject: Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download Citeren Dan [EMAIL PROTECTED]: However, is there an option so that the sound-dialog can be forced to just use the defaults it preselected for me ? At first run, you are asked to select the audio devices... one time only.. Then it will be used at each run Yes, this was clear to me. However the defaults are usually fine - and if they are not, the person will most likely know about it because he has to manually configure anything that uses soundcards :-)) I'd be fine with it just accepting defaults if something was found to be out of order.. I'd be happy to know if others have different views on this though... Ok... you're right. I'll make it to take the default sound device for Rec and Play and PC Speaker for ring, without asking to configure it at first run. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
i believe the more accepted term is 'basics' as in asterisk-basics Grzegorz Nosek wrote: On Thu, 20 Nov 2003 14:54:15 -, Linus Surguy wrote So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) Maybe asterisk-install ? asterisk-starters ? ___ 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] New DIAX - version 0.9.4 - a big step forward - available for download
i think that is a bad idea, atm i have the option to use my screen speakers for ringing and my headset for the actual audio. My pc speaker sux bigtime (too quiet) but i agree that putting an option for the pc speaker is a good idea. Zoa. Ok... you're right. I'll make it to take the default sound device for Rec and Play and PC Speaker for ring, without asking to configure it at first run. Best regards, Dan ___ 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] Bayonne and Asterisk
I never used Bayonne as a PBX of any kind, only for IVR, that's what it was designed for. You can put other packages with Bayonne to get it to work with some VOIP protocols supposedly , but it would be much more work to do that than to just set up Asterisk. MATT--- -Original Message- From: Uriel Carrasquilla [mailto:[EMAIL PROTECTED] Sent: Sunday, November 23, 2003 1:06 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Bayonne and Asterisk How about issues such as echo, voice quality, supported codec's? does it work with SIP? Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of mattf Sent: Tuesday, November 18, 2003 9:46 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Bayonne and Asterisk I used Bayonne for 2 years before switching to Asterisk. Right now I'm still running Bayonne on one application and it's been running happily without me looking at it for over 6 months. I'd say these are the strengths of Bayonne: - Runs on Dialogic, Pika and other widely available hardware - extremely reliable, mine never crashes and here are the weaknesses: - nowhere near as active of a support community as Asterisk has - configuration of the hardware/drivers is a nightmare compared to Asterisk/Digium - it is quite limited in it's included apps, IVR and voicemail - not as many options for scripting as Asterisk - it was not designed to have full PBX functionality, some PBX functionality is added as afterthought - the code/organization/flow is not as well thought out or documented as Asterisk is And yes, they can run fine together(I'm not using VOIP, just a T1 out of Asterisk to Bayonne to test and see if it would work). The IVR application that I currently still have running on Bayonne is only still on Bayonne because it can never go down, and Bayonne has proven itself to me to be extremely stable, while I cannot personally say AT THIS TIME that an Asterisk box would stay up for over 6 months with no crashes. At last check I was never able to get VOIP inbound working on Bayonne, maybe this has changed in the last 6 months but if you do get it working I'd be interested to find out how. MATT--- -Original Message- From: Dirk-Jan Wemmers [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 18, 2003 8:44 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Bayonne and Asterisk All, is anyone using Bayonne in conjunction with Asterisk? I'm currently using only Bayonne, but I'm investigating the possibilities of switching the telephony frontend over to Asterisk, and have Asterisk route the IVR tasks to Bayonne through H323. Anyone care to share his views on this approach? Any pointers or do's and don'ts? All info is greatly appreciated! Regards, Dirk-Jan -- Dirk-Jan Wemmers, Capcave B.V. Zonnebaan 17, 3542EA Utrecht T +31(0)30-2149670, F +31(0)30-2149679 M +31(0)651 063040, E [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 ___ 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] Stutter dialtone but no messages
On Sat, 22 Nov 2003, marrandy wrote: Now, I am getting the stutter/dialtone, that tells me I have a message, But when I check both the new and old messages on each of the 3 mailbox accounts, there are no messages. I had this problem the other day. I think it was caused by me trying to pick up a message whilst it was still being recorded. The incoming call I was listening to was cut off, and it also left behind a file in the INBOX. With that file present it continued to stutter the dialtone. After I deleted that one file it stopped. The file just had one line which I think was the time length of the message. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Feedback with X100P and SIP fwd.pulver
Hello: I have installed *. I configured my SIP account and my X100P. But whenI call from SIP or from PSTN. The SIP extension hear an echovoice of its conversation. Anyone can help me??? Thanks, voipfan
Re: [Asterisk-Users] Feedback with X100P and SIP fwd.pulver
Ya learn to search the archives. This has been covered MANY MANY times. bkw On Sun, 23 Nov 2003, VoIP Fan wrote: Hello: I have installed *. I configured my SIP account and my X100P. But when I call from SIP or from PSTN. The SIP extension hear an echo voice of its conversation. Anyone can help me??? Thanks, voipfan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo Cancellation, TDMoE fails, X100P works
System Diagram: +- PRI -- modem bank | LEC -- PRI -- * with --+ T400P --+ | (pri) +- TDMoE -- * with -- SIP -- sip phones LEC -- analog line --- X100P PS The modem connections have been working just fine after some initial problems and a CVS update. Thanks Mark! Would you be willing to share your T400P/NAS configuration? I intend on doing something very similar but was told that the T400P couldn't pass traffic between the LEC PRI and the NAS and achieve v.90/v.92 connections. Oh, but it DEFINATELY can! # file: /etc/zaptel.conf (on * with T400P) loadzone=us # pri from LEC span=1,1,0,esf,b8zs # pri T1 to portmaster (this one is online) span=2,0,0,esf,b8zs # pri T1 to portmaster (backup, testing) span=3,0,0,esf,b8zs # pri T1 (unused, next I will try using this to # connect to the other * box if I can't stop the # echos on TDMoE) span=4,0,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 # For Testing: This maps channels straight across with as little involvement # as possible by * (the zaptel driver just shuffles bits). # dacs=1-24:25 bchan=49-71 dchan=72 bchan=73-95 dchan=96 # TDMoE channels to PBX (* with X100P's) dynamic=eth,eth1/00:01:02:9B:39:72,24,1 bchan=97-119 dchan=120 # file: /etc/asterisk/zapata.conf [channels] language=en context=lec-pri signalling=pri_cpe switchtype=5ess usecallerid=yes group=1 echocancel = no echocancelwhenbridged = no channel = 1-23 context=outbound-pm signalling=pri_net switchtype=5ess usecallerid=yes group=2 echocancel = no echocancelwhenbridged = no channel = 25-47 context=outbound-pm signalling=pri_net usecallerid=yes group=3 channel = 49-71 context=local switchtype=5ess signalling=pri_net usecallerid=yes group=4 channel = 73-95 ;; TDMoE to PBX context=local usecallerid=yes signalling=pri_net group=5 channel = 97-119 # file: /etc/asterisk/extensions.conf (excerpt) [lec-pri] ; these numbers to our portmasters exten = 5413452121,1,Dial(Zap/g2/5413452121) exten = 5413452122,1,Dial(Zap/g2/5413452122) ; these DID's to PBX via TDMoE exten = 5413452123,1,Dial(Zap/g5/5413452123) Note the commented out line in zaptel.conf (# dacs=1-24:25). This was new in CVS as of a couple of weeks ago and this was the first thing that worked successfully with the 56k modem traffic. Also note that echo canceling is off in zapata.conf, it could only confuse the modems. Also, either * is _very_ fast when it shuffles bits between channels and/or modems have some decent echo canceling software builtin, because the modems operate at full speed. Also note that neither the 'c' option nor the 'd' option is used in the Dial command in extensions.conf. In all of our tests before talking to Mark we had used those options (seemed the obvious thing to do judging from their descriptions). Mark suggested taking them out, we did, it worked. (I don't know if that is the whole difference between what worked and what did not. This was one of those situations where you go in thinking the solution is straight forward, but after 6 or a dozen failed attempts you realize that you are in for a big trial and error session and you have not kept exacting records on the initial failed attempts.) Let me know if that helps. Now, it seems to me that the only reason you would want to do what I did is so that you can route some of the incoming calls somewhere else for another purpose, like a voice call. And somewhere along the line you will run into my problem: echo cancellation. Maybe you can help me out there. Also, your diagram looks like you're connecting two * boxes by plugging one end of a cable into an ethernet jack, and the other into a T400P jack -- is this correct? I thought TDMoE was strictly ethernet--ethernet? I left out some details for the sake of brevity. Both * boxes have 2 ethernet interfaces, one for normal IP and the other dedicated to the TDMoE traffic. Regards, Andrew Gary ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback with X100P and SIP fwd.pulver
Well, SIP to SIP with no intervening analogue should produce no echo at all. Echo on SIP to analogue calls has been covered extensively on this list. Do a search on echo. Iain Hello: I have installed *. I configured my SIP account and my X100P. But when I call from SIP or from PSTN. The SIP extension hear an echo voice of its conversation. Anyone can help me??? Thanks, voipfan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback with X100P and SIP fwd.pulver
Hey Brian... when resetting transmission levels in zapata.conf, is restarting * enough? Ya learn to search the archives. This has been covered MANY MANY times. bkw On Sun, 23 Nov 2003, VoIP Fan wrote: Hello: I have installed *. I configured my SIP account and my X100P. But when I call from SIP or from PSTN. The SIP extension hear an echo voice of its conversation. Anyone can help me??? Thanks, voipfan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup ???
I was wondering if you can pick up a ringing channel by dialing *8# when you and the other phones are in pickupgroup. Could you do something to the effect of If the caller was put on a certain extension and just sitting there... Could you grab the caller by doing something like *8exten where the caller is sitting? Here is my issue. I need to find a way to put the caller on hold and play MOH. I don't want to use park unless I can park them on a given extension. And if I use flash the user gets a dial tone. He can either transfer to another user at that point, and if he hangs up the line rings back until he picks up. Any ideas ?? -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
On Sat, 2003-11-22 at 21:25, [EMAIL PROTECTED] wrote: This book will be available in electronic form under some sort of open publishing license, in addition to being sold in bookstores, right? Yes, that's the plan. I personally am a lot more interested in this from the standpoint of having electronic documentation. Having a nice dead-tree book would be great and wonderful, but it's not my primary focus. I haven't decided on the exact details, but the book will be written under some kind of open publishing license. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
On Sat, 2003-11-22 at 21:22, [EMAIL PROTECTED] wrote: please please please if you are going to write something like that, write it using something like texinfo or groff or docbook or whatever so that you can make it available in a wide variety of formats. you should not have to run non-free software in order to be able to search the documentation (and no, acroread is not free software) It's being done in DocBook (at least for now) just because that's what I happen to know a little about. Plus, it makes it simple to create HTML, PDF, etc. versions from the source file. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax
Hi Dan, From the log, it looks like something wrong is in the modem software. I have been adding a V.27ter modem to the FAX software, to do the slow 2400, and 4800 bps FAX modes. I've been busy with other things the last couple of weeks, but I am finishing this off now. I have also made the existing V.29 modem more robust on bad lines. When I get this released (should be this week) I will look into the issues that you and some others are having. It seems the current software works OK with some FAX machines and not others. Perhaps I need to make it more tolerant. :-) Regards, Steve Dan Fernandez wrote: I am also having problems receiving my first fax. I get a 320byte file (for a 4 page fax). If I look a the tiff generated, is just has some few dots. I am sending the fax from a notebook with Windows XP to an X100P and using libtiff v3.5.7. Has anyone successfully received faxes ? Output to the console as follows: Changed from phase 0 to 1 Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Store and forward Internet fax: no Real-time Internet fax: no Preferred octets: 256 Can receive fax Data signalling rate: V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Inch-based resolution preferred: no Metric-based resolution preferred: no Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 c6 f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 TSI: 43 20 20 20 20 20 37 39 36 39 36 33 32 20 31 31 39 20 39 34 2b TSI without final frame tag Remote fax gave TSI as: +49 911 2369697 DCS: 83 00 46 f0 00 DCS with final frame tag In state 9 DCS: Store and forward Internet fax: no Real-time Internet fax: no Can receive fax Data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Get at V.29 Changed from phase 3 to 5 Fast carrier up Fast carrier down Changed from phase 5 to 4 0 bad bits in trainability test Start rx document - compression 1 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Equalizer state: -7 (0.0, 0.0) - 0.0 -6 (0.0, 0.0) - 0.0 -5 (0.0, 0.0) - 0.0 -4 (0.0, 0.0) - 0.0 -3 (0.0, 0.0) - 0.0 -2 ( -0.08332,-0.68161) - 0.47154 -1 (0.66136, 0.74688) - 0.99522 0 (0.83336, 2.86588) - 8.90777 1 (0.66136, 0.74688) - 0.99522 2 ( -0.08332,-0.68161) - 0.47154 3 (0.0, 0.0) - 0.0 4 (0.0, 0.0) - 0.0 5 (0.0, 0.0) - 0.0 6 (0.0, 0.0) - 0.0 7 (0.0, 0.0) - 0.0 Equalizer state: -7 (0.24872,-0.01973) - 0.06225 -6 (0.06295,-0.59891) - 0.36265 -5 (0.04198,-0.41768) - 0.17622 -4 (0.12922, 0.34088) - 0.13290 -3 (0.29121, 0.41776) - 0.25932 -2 (0.05779,-1.18501) - 1.40758 -1 (0.67094,-0.35069) - 0.57314 0 (0.60670, 2.23125) - 5.34655 1 (0.34490, 1.36992) - 1.99564 2 ( -0.31233, 0.09159) - 0.10594 3 ( -0.00825, 0.00787) - 0.00013 4 (0.02226,-0.47177) - 0.22306 5 ( -0.17953, 0.14528) - 0.05334 6 ( -0.25318, 0.57800) - 0.39819 7 ( -0.10465, 0.05659) - 0.01415 Equalizer state: -7 (0.10674, 0.25606) - 0.07696 -6 ( -0.05000,-0.10084) - 0.01267 -5 ( -0.04246,-0.24755) - 0.06308 -4 (0.01990, 0.30283) - 0.09210 -3 ( -0.04673, 0.31177) - 0.09939 -2 ( -0.20575,-0.84531) - 0.75687 -1 (0.53295, 0.18096) - 0.31678 0 (0.84089, 2.50094) - 6.96180 1 (0.69048, 1.16134) - 1.82547 2 ( -0.06589,-0.16464) - 0.03145 3 ( -0.04671, 0.30535) - 0.09542 4 (0.07891,-0.09615) - 0.01547 5 (0.10829,
Re: [Asterisk-Users] RxFax
existing V.29 modem more robust on bad lines. When I get this released (should be this week) I will look into the issues that you and some others are having. It seems the current software works OK with some FAX machines and not others. Perhaps I need to make it more tolerant. :-) I have a couple of fax machines I will test the code with once you're done what you're doing... they are notoriously bad... A Canon ImageRunner 3300 and whatever they have at our parent company in the U.S.. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk] GSM access
Hi Mark Did you or anyone else ever find a satisfactory solution to this? Are there any phones which provide voice through the serial connection? What about the nokia card phone - does it have open source drivers? Cheers Rob On Sat, Jul 13, 2002 at 10:31:53AM -0500, Mark Spencer wrote: Does anyone (maybe in Europe) know how I could build a GSM compatible channel for Asterisk, so that one could call other mobile phones from Asterisk, or build a portable phone system, with GSM channels being used for outside access? Is there any hardware for PC's or a way to rig up a phone with a serial connection and a sound card to use it? Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Express Router Asterisk
Greetings... We've been having some interoperability issues between Asterisk and an AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 somewhere. So, I've been pondering using iptel.org's SIP server (SIP Express Router) as a front end for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or anything at all involving SER and Asterisk? Thanks! -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agi exec problem.
hi folks. (apologies in advance if this is a particularly stupid question) just getting my feet wet with asterisk / agi, and am a little stuck using EXEC. it works fine for applicaitons that take simple arguments, but chokes on applications that require multiple words as arguments. for example, this works fine: EXEC Playback(demo-congrats) but this doesn't: EXEC add extension s,3,Playback(demo-congrats) into local problem seems to be that AGI reads the second example to be: EXEC add extension and ignores the rest (presumably because it assumes the space after 'extension' singifies the end of the argument) is there a way around this? thanks, tad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk] GSM access
Robert Murray wrote: Hi Mark Did you or anyone else ever find a satisfactory solution to this? Are there any phones which provide voice through the serial connection? What about the nokia card phone - does it have open source drivers? Cheers Rob On Sat, Jul 13, 2002 at 10:31:53AM -0500, Mark Spencer wrote: Does anyone (maybe in Europe) know how I could build a GSM compatible channel for Asterisk, so that one could call other mobile phones from Asterisk, or build a portable phone system, with GSM channels being used for outside access? Is there any hardware for PC's or a way to rig up a phone with a serial connection and a sound card to use it? Mark It seems to me that the Nokia 32 GSM terminal would be the best bet. Has anyone used one yet with Asterisk? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phone compatibility list
Hello! Is there a phone compatibility list anywhere? I know Asterisk is supposed to be compatible with IP phones that support SIP, h.232, and IAX, but a list of phone known to be supported would be a nice addition to the documentation. I need a buisiness phone, and the Cisco is _EXPENSIVE_ I like the 3com 1102, but it is NBX phone, and I can find no documentation of this phone working with Asterisk, even though it is listed as a SIP phone, and is supposed to support h.232. Anyone got this one working? Michael Rowley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agi exec problem (followup)
actually, i do have a workaround which bypasses the exec command entirely: system(asterisk -r -x 'add extension s,3,Playback(demo-congrats) into local'); but it's ugly. seems like it should be possible to do this with exec. .t -- Forwarded message -- Date: Sun, 23 Nov 2003 21:17:50 -0500 (EST) From: tad [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: agi exec problem. hi folks. (apologies in advance if this is a particularly stupid question) just getting my feet wet with asterisk / agi, and am a little stuck using EXEC. it works fine for applicaitons that take simple arguments, but chokes on applications that require multiple words as arguments. for example, this works fine: EXEC Playback(demo-congrats) but this doesn't: EXEC add extension s,3,Playback(demo-congrats) into local problem seems to be that AGI reads the second example to be: EXEC add extension and ignores the rest (presumably because it assumes the space after 'extension' singifies the end of the argument) is there a way around this? thanks, tad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] agi exec problem.
On Sunday 23 November 2003 20:17, tad wrote: hi folks. (apologies in advance if this is a particularly stupid question) just getting my feet wet with asterisk / agi, and am a little stuck using EXEC. it works fine for applicaitons that take simple arguments, but chokes on applications that require multiple words as arguments. for example, this works fine: EXEC Playback(demo-congrats) but this doesn't: EXEC add extension s,3,Playback(demo-congrats) into local problem seems to be that AGI reads the second example to be: EXEC add extension and ignores the rest (presumably because it assumes the space after 'extension' singifies the end of the argument) is there a way around this? You're confusing the CLI with applications. You can _only_ EXEC an application. See the CLI command 'show applications' for a list of the applications you can EXEC. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] agi exec problem.
asterisk*CLI show agi answer Asserts answer wait for digit Waits for a digit to be pressed send text Sends text to channels supporting it receive char Receives text from channels supporting it tdd mode Sends text to channels supporting it stream file Sends audio file on channel send image Sends images to channels supporting it say digits Says a given digit string say number Says a given number get data Gets data on a channel set context Sets channel context set extension Changes channel extension set priority Prioritizes the channel record file Records to a given file set autohangup Autohangup channel in some time hangup Hangup the current channel exec Executes a given Application set callerid Sets callerid for the current channel channel status Returns status of the connected channel set variable Sets a channel variable get variable Gets a channel variable verbose Logs a message to the asterisk verbose log database get Gets database value database put Adds/updates database value database del Removes database key/value database deltree Removes database keytree/value noop Does nothing set music Enable/Disable Music on hold generator I don't see add extension in the list of AGI commands. bkw On Sun, 23 Nov 2003, tad wrote: hi folks. (apologies in advance if this is a particularly stupid question) just getting my feet wet with asterisk / agi, and am a little stuck using EXEC. it works fine for applicaitons that take simple arguments, but chokes on applications that require multiple words as arguments. for example, this works fine: EXEC Playback(demo-congrats) but this doesn't: EXEC add extension s,3,Playback(demo-congrats) into local problem seems to be that AGI reads the second example to be: EXEC add extension and ignores the rest (presumably because it assumes the space after 'extension' singifies the end of the argument) is there a way around this? thanks, tad ___ 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] agi exec problem (followup)
What is the goal of this? It doesn't make much sense to me. Care to share some insite into what your goal is? bkw On Sun, 23 Nov 2003, tad wrote: actually, i do have a workaround which bypasses the exec command entirely: system(asterisk -r -x 'add extension s,3,Playback(demo-congrats) into local'); but it's ugly. seems like it should be possible to do this with exec. .t -- Forwarded message -- Date: Sun, 23 Nov 2003 21:17:50 -0500 (EST) From: tad [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: agi exec problem. hi folks. (apologies in advance if this is a particularly stupid question) just getting my feet wet with asterisk / agi, and am a little stuck using EXEC. it works fine for applicaitons that take simple arguments, but chokes on applications that require multiple words as arguments. for example, this works fine: EXEC Playback(demo-congrats) but this doesn't: EXEC add extension s,3,Playback(demo-congrats) into local problem seems to be that AGI reads the second example to be: EXEC add extension and ignores the rest (presumably because it assumes the space after 'extension' singifies the end of the argument) is there a way around this? thanks, tad ___ 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
[Asterisk-Users] CVS update of Asterisk - What did I do wrong?
Late Sunday night, getting cvs update asterisk ? asterisk/doc/api cvs server: Updating asterisk M asterisk/app.c cvs [server aborted]: missing expected branches in /usr/cvsroot/asterisk/asterisk-ng-doxygen,v [EMAIL PROTECTED] src]# Checkout does same thing What did I mess up? __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS update of Asterisk - What did I do wrong?
Works fine from here... blow your src tree away and start fresh. bkw On Sun, 23 Nov 2003, Jonathan Biggs wrote: Late Sunday night, getting cvs update asterisk ? asterisk/doc/api cvs server: Updating asterisk M asterisk/app.c cvs [server aborted]: missing expected branches in /usr/cvsroot/asterisk/asterisk-ng-doxygen,v [EMAIL PROTECTED] src]# Checkout does same thing What did I mess up? __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ 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] agi exec problem.
On Sun, 2003-11-23 at 21:30, Brian West wrote: asterisk*CLI show agi answer Asserts answer wait for digit Waits for a digit to be pressed send text Sends text to channels supporting it receive char Receives text from channels supporting it tdd mode Sends text to channels supporting it stream file Sends audio file on channel send image Sends images to channels supporting it say digits Says a given digit string say number Says a given number get data Gets data on a channel set context Sets channel context set extension Changes channel extension set priority Prioritizes the channel record file Records to a given file set autohangup Autohangup channel in some time hangup Hangup the current channel exec Executes a given Application set callerid Sets callerid for the current channel channel status Returns status of the connected channel set variable Sets a channel variable get variable Gets a channel variable verbose Logs a message to the asterisk verbose log database get Gets database value database put Adds/updates database value database del Removes database key/value database deltree Removes database keytree/value noop Does nothing set music Enable/Disable Music on hold generator I don't see add extension in the list of AGI commands. Yes, but AGI does have the exec command that was needed. Then exec takes arguments. On Sun, 23 Nov 2003, tad wrote: hi folks. (apologies in advance if this is a particularly stupid question) just getting my feet wet with asterisk / agi, and am a little stuck using EXEC. it works fine for applicaitons that take simple arguments, but chokes on applications that require multiple words as arguments. for example, this works fine: EXEC Playback(demo-congrats) but this doesn't: EXEC add extension s,3,Playback(demo-congrats) into local problem seems to be that AGI reads the second example to be: EXEC add extension and ignores the rest (presumably because it assumes the space after 'extension' singifies the end of the argument) is there a way around this? thanks, tad -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nufone account not registering
hi, my * box is behind a NAT. i am using netgear router. it seems that my toll free number is unable to register with nufone because my * box is behind NAT firewall. the outgoing calls are working fine and nice. what can be done? cm configurations are fine. = Designs __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone account not registering
This is not a private issue as far as i know. u misundersood me. the nufone account is fine. the real problem is with the asterisk NAT issue. i was asking for help if any one had similar problem with nufone account. i am using IAX. is there anything like nat=yes as in sip.conf?? i read iax should work with normal configuration. its ok with outbound. i only have problems with inbound. hope u understood what i am trying to say. thank you. cm --- Tilghman Lesher [EMAIL PROTECTED] wrote: On Sunday 23 November 2003 23:46, C M wrote: my * box is behind a NAT. i am using netgear router. it seems that my toll free number is unable to register with nufone because my * box is behind NAT firewall. the outgoing calls are working fine and nice. what can be done? When you're having trouble with a particular service, especially one that supports your configuration, I would think both they and we would appreciate it if you took up the matter privately with the support department of that service, before you make it a public issue. In this case, it's NuFone, which does support Asterisk. Instead of posting to this public mailing list, which is not NuFone-specific, please contact the fine folks at NuFone, as they will likely be more than happy to help you. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download
Hi, - Original Message - From: zoa [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, November 23, 2003 8:32 PM Subject: Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download i think that is a bad idea, atm i have the option to use my screen speakers for ringing and my headset for the actual audio. My pc speaker sux bigtime (too quiet) but i agree that putting an option for the pc speaker is a good idea. Zoa. Ok... you're right. I'll make it to take the default sound device for Rec and Play and PC Speaker for ring, without asking to configure it at first run. The problem is that as you manually configure the devices, it will remain like that, so... what's set at the first run is only for the cases when the device from the config file does not exist on the local system. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
