Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
On Tuesday 14 October 2003 23:50, Chris Albertson wrote: > If the software needs a specialcard to keep time then the > software is broken or poorly designed. Don't complain so loudly unless you're willing to contribute the fixes. Opinions are like assholes, and you know where that's going. Takes something else entirely to fix a perceived problem. If the system is so horribly broken, why are you using it as-is and not fixing it? -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec
Steven J. Sobol wrote: X-Lite build 1079 consistently chokes no matter which codec I use - after five seconds I suddenly have no sound coming in and possibly no sound going out too. Putting the line I'm on on hold and then switching back to it gives me another five seconds of sound, then it dies, etc. If you are not using a headset then X-Lite cuts the sound after a few seconds.. Its probably the echo cancelation kicking in.. Connect a headset and all should be fine.. Later. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DISA and ringing tone
Hi I am using DISA to get my Polycom SoundPoint400 with H323 firmware to connect to * I have it working, but when I dial SIP end points there is no ringing tone on the phone. DISA gives dial tone but does not give ringing (if I understand correctly it is because it expects to transmit sound created by terminating side of the call) Is there a way to make DISA application to generate ringing tone back to the handset of the originating end point? Thanks, Serge Serge - I don't know about H323, but I get ringtones from DISA on a SIP outbound channel. Try adding "r" to the options list on your Dial statement. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] e100p in Australia
I've seen this question asked before but haven't seen a definative answer. Does the e100p work in australia? Did any one who was asking the question before bite the bullet and get one? I can get a te410 if i really have to but would prefer to stay with the cheaper option. Any comments appreciated, Thanks - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildcard TDM400P - FXO?
On Tue, 2003-10-14 at 19:31, Gene Kochanowsky wrote: > Does anyone know if or when the FXO daughter boards for the TDM400P will be > available? September 2003. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec
Do you have a 100 or 101? You have indicated different models in your postings. Were you able to get Call Transfer and Call Waiting working with your Asterisk system and other phones? Which version of the Grandstream firmware do you use? There most recent on their website this weekend was at least 2 version numbers higher than what came on my phone in August. Think that they are making improvements rather frequently. Robert > On Wed, 15 Oct 2003, Jon Pounder wrote: > >> >The Grandstream 101 I'm using is a piece of junk but I don't have the >> same >> >problem with it. >> >> What don't you like about the grandstream ? (I am not looking to flame >> you, >> but was considering buying and if there are problems would rather find >> out >> beforehand) > > Nothing works. Call transfer and call waiting, in particular. (Well, > almost nothing; vm notification does work) > > There is no place to plug in a headset, and since I do a fair amount of > tech support and longish conference calls, that's a big deal for me. > > However, keep in mind that I have an old, no-longer-manufacturered model > (the Budgetone 100). Don't take my frustration with my outdated phone as > a sign that you should dismiss Grandstream out of hand - I just don't like > my 100. > > -- > JustThe.net Internet & Multimedia Services > 22674 Motnocab Road * Apple Valley, CA 92307-1950 > Steve Sobol, Proprietor > 888.480.4NET (4638) * 248.724.4NET * [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] Digium should develop and sell just Dummy card. For timing...
From: "Anton Tinchev" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, October 15, 2003 1:06 PM Subject: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing... > I'm first to buy 5 pack. Even for > $30. > Doesn't ztdummy already do this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP authentication
Yes Sean It looks fine with 200 Responses, but actually it doesnt ring while dialing from other phone. Might this be the problem of SIP UA? if i try X-Lite that continously goes in loop to register(One bad thing with X-Lite is that it uses transparent proxy's IP as its own., so never gets reply from asterisk, going in loop of register process) Secondly i tried to live with that "Acquired" message, and using extension.conf, then the dialing UA gets message "Established" and Terminates media channel in 2,3 seconds. While there is no actual activity on dialed UA strange..:) JF"Sean P. Robertson" <[EMAIL PROTECTED]> wrote: It looks like you are registering fine. If you dial 12321 from another phone, does it not ring? This is the transaction as I see it in the log that you attached: Phone: REGISTER Asterisk: Proxy Authentication Required (Send me your credentials) Phone: REGISTER with CREDENTIALS Asterisk: 200 OK (You are now registered) Asterisk: NOTIFY (You have 0/0 messages in your voicemail.) Phone: 200 OK (Thanks for letting me know) Sean ___ Sean Robertson NETXUSAp. 800-289-6389f. 864-233-4344 "Ask me about Voice over IP."http://www.netxusa.com/ - Original Message - From: John Foster To: [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 12:49 AM Subject: [Asterisk-Users] Problem with SIP authentication Hi List, After going through mailing list and manual of asterisk, I still could not properly get authenticated with my SIP UA to asterisk. i m using a username for UA "12321" and following are SIP.conf file user params [12321]type=friendusername=12321host=dynamicsecret=ccartacontext=defaultmailbox=1234,2345 ; Mailbox for message waiting indicator [7]type=friendusername=7host=dynamicsecret=atracccontext=defaultmailbox=1234,2345 m trying with SIPPS UA, that gets status "Acquired", not "Registered", Can anyone give any idea about it? I tried same with X-Lite, didnt work. Sip debug messages are pasted below. Best Regards, JF Sip read:REGISTER sip:192.168.100.71 SIP/2.0Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: ;tag=3b2cf0baTo: Call-ID: [EMAIL PROTECTED]Contact: ccarta ;expires=600;q=0.500Expires: 600CSeq: 1 REGISTERContent-Length: 0User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 10 headers, 0 linesUsing latest request as basis requestSending to 192.168.100.66 : 5062 (non-NAT)Transmitting (no NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: ;tag=3b2cf0baTo: ;tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: Content-Length: 0 to 192.168.100.66:5062Transmitting (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: ;tag=3b2cf0baTo: ;tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: Proxy-Authenticate: Digest realm="asterisk", nonce="7c7fba4d"Content-Length: 0 to 192.168.100.66:5062Sip read:REGISTER sip:192.168.100.71 SIP/2.0Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: ;tag=3b2d0018To: Call-ID: [EMAIL PROTECTED]Contact: ccarta ;expires=600;q=0.500Expires: 600CSeq: 2 REGISTERContent-Length: 0Proxy-Authorization: Digest username="12321",realm="asterisk",uri="sip:192.168.100.66",nonce="7c7fba4d",nc="0001",response="b8b1d7fc53eff354dfc31dfa3f800749"User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 11 headers, 0 linesUsing latest request as basis requestSending to 192.168.100.66 : 5062 (non-NAT)Transmitting (no NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: ;tag=3b2d0018To: ;tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: Content-Length: 0 to 192.168.100.66:5062Transmitting (no NAT):SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: ;tag=3b2d0018To: ;tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERExpires: 600Contact: ;expires=600Date: Tue, 14 Oct 2003 13:46:14 GMTContent-Length: 0 to 192.168.100.66:506211 headers, 2 linesReliably Transmitting:NOTIFY sip:[EMAIL PROTECTED]:5062 SIP/2.0Via: SIP/2.0/UDP 192.168.100.71:5060;branch=z9hG4bK3ecb7a3bFrom: "asterisk" ;tag=as3f6e8c0eTo: Contact: Call-ID: [EMAIL PROTECTED]CSeq: 102 NOTIFYUser-Agent: Asterisk PBXEvent: message-summaryContent-Type: application/simple-message-summaryContent-Length: 36 Messages-Waiting: noVoicemail: 0/0 (no NAT) to 192.168.100.66:5062Sip read:SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.100.71;branch=z9hG4bK3ecb7a3bF
Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
If the software needs a specialcard to keep time then the software is broken or poorly designed. I've written software to control a kind of "video" camera. (actualy "TDI" or "drift scan" that takes still pictures of moving abojects.) that worked at the very low level and in software ganerated the timming pulses that went to the lead pins on the CCD sensor that moved charge around on the CCD chip. It's what you call "hard" real time down at the usec level. This ran on a 486 (or higher) and took advantage of the CPU's built-in timmer's ability to generate interrupts at a specified rate. There are other ways if you have a Pentium. The details don't matter. But be assured that no extra hardware should be required. There are mutiple non-hardware solutions available. --- Anton Tinchev <[EMAIL PROTECTED]> wrote: > I'm first to buy 5 pack. Even for > $30. > > ___ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec
On Wed, 15 Oct 2003, Jon Pounder wrote: > >The Grandstream 101 I'm using is a piece of junk but I don't have the same > >problem with it. > > What don't you like about the grandstream ? (I am not looking to flame you, > but was considering buying and if there are problems would rather find out > beforehand) Nothing works. Call transfer and call waiting, in particular. (Well, almost nothing; vm notification does work) There is no place to plug in a headset, and since I do a fair amount of tech support and longish conference calls, that's a big deal for me. However, keep in mind that I have an old, no-longer-manufacturered model (the Budgetone 100). Don't take my frustration with my outdated phone as a sign that you should dismiss Grandstream out of hand - I just don't like my 100. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec
The Grandstream 101 I'm using is a piece of junk but I don't have the same problem with it. What don't you like about the grandstream ? (I am not looking to flame you, but was considering buying and if there are problems would rather find out beforehand) -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [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] Eicon Diva Server BRI (T1) Cards
On Wed, Oct 15, 2003 at 12:02:18AM +0200, Roger Schreiter wrote: > Hi, > > my asterisk experiences with isdn cards supported by i4l Out of interest, which cards. > are not very good, but with avm a1 and capi everything > works very fine and stable. (SuSE Linx 8.2, Kernel 2.4.20, > german ISDN). Heard good things about capi and AVM, but unfortunately none are approved for Australia :-( > Now I want to connect a T1. Should I use an AVM T1-B > for approx 6000 EUR or is it ok to use one of Eicon's cheaper > Diva Server BRI S2M cards? T1 = ISDN PRI in the US/Japan (24 lines) ISDN BRI = 2 lines So I don't think that card would work, you want a US PRI card if they make one. If you are in Germany, I think you want E1 = ISDN PRI outside US/Japan (30 lines) and for that you want a E100P for US$595 from digium. > Do they use i4l or is there a similar thing as AVM's capi > driver for eicon's S2M cards? I think there is a channel driver for the eicon, but I don't know where from. > Are there any experiences how they compare when used > with asterisk? I am interested in this too, the Eicon cards are also certified for Australia. cheers, -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec
X-Lite build 1079 consistently chokes no matter which codec I use - after five seconds I suddenly have no sound coming in and possibly no sound going out too. Putting the line I'm on on hold and then switching back to it gives me another five seconds of sound, then it dies, etc. The Grandstream 101 I'm using is a piece of junk but I don't have the same problem with it. Not sure whether the issue is X-Lite or the Asterisk server. Has anyone else had this problem? -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VAD in Asterisk ?
Hi, Is there is some form of VAD on * for SIP channels, cause I have a problem with MOH. I made an extension which simply plays MOH, when I dial that extension with my ATA188 MOH sounds choppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don´t have this kind of problem on my Cisco 7960 which has VAD deactivated. The problem I don't see any VAD option in AudioModes of ATA. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DISA and ringing tone
Hi I am using DISA to get my Polycom SoundPoint400 with H323 firmware to connect to * I have it working, but when I dial SIP end points there is no ringing tone on the phone. DISA gives dial tone but does not give ringing (if I understand correctly it is because it expects to transmit sound created by terminating side of the call) Is there a way to make DISA application to generate ringing tone back to the handset of the originating end point? Thanks, Serge _ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WCFXO echo rexolved for me
Hello, I resolved my echo issue using grandstream/estara etc etc sip phones and wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl motherboard for an asus p4pe? i845? based motherboard and the echo has completly gone away along with aggressive suppressor option in the makefile. I hope this helps others. Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH 43017 Phone: 614-798-9106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium cards just for timing - moden cards?
most probably use tjnetworks chips, www.tjnet.com to be specific: http://www.tjnet.com/chips/tiger320.htm I am not sure though:) - Original Message - From: "Leo Ann Boon" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, October 15, 2003 6:09 AM Subject: Re: [Asterisk-Users] Digium cards just for timing - moden cards? > I think the X101P uses an Ambient/Intel HAM 56K modem chip. The older > X100 is based on Motorola. Any authoritative answers? > > > Andrew Kohlsmith wrote: > > >>Is that why there is an X100P and an X101P? What design is the X101P > >>based on? > >> > >> > > > >AFAIK the current design uses a Tiger 320 chip which is essentially a PCI > >gateway -- it provides a serial port and an 8-bit parallel interface to > >anything. The single FXO card uses the serial interface, and the single T1 > >card uses the 8-bit parallel interface to a standard Dallas Semiconductor > >T1/E1/J1 framer IC. > > > >Regards, > >Andrew > >___ > >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] Digium should develop and sell just Dummy card. For timing...
I'm first to buy 5 pack. Even for > $30. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outbound caller ID problem on PRI
I can't seem to hide and/or set my caller ID from *.
I'm using a quite recent (three weeks or so) CVS with an E400P card.
I have pridialplan=unknown in zapata.conf and I'm based in the UK.
The relevant bit of pri debug looks like this (reformatted to fit 80
char width):
> Calling Number (len= 4) [ Ext: 0
> TON: Unknown Number Type (0)
> NPI: Unknown Number Plan (0)
> Presentation: Unknown (67) '' ]
I'm dialing in from SIP outbound to Zap with a context like this:
exten => _X.,1,SetCallerID("mxtelecom" <0845123456>)
exten => _X.,2,Dial(Zap/1/${EXTEN})
Although:
exten => _X.,1,SetCallerID(0845123456|a)
exten => _X.,2,Dial(Zap/1/${EXTEN})
Equally doesn't work.
I've tried setting these in zapata.conf:
callerid="foo" <0845123456>
hidecallerid=yes
No matter what I do, I get a default caller ID provided by my telco. If
I prefix the number dialed with 141 (standard UK hide caller ID) the
caller ID isn't presented to the end user, but this is an ugly kludge
that I don't want to have to do. Ideally, I'd like to be able to set my
callerID to an arbitrary number.
If I set pridialplan=national/international I can't work out what format
the outbound calls numbers should take and get denied messages back.
Anyone have any ideas?
--
Alastair Maw
MX Telecom - Systems Analyst
www.mxtelecom.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line in use detection...
> > I would like to know if is possible to setup my Asterisk to detect if the > > phone lines from FXO cards are in use. We use the parallel phones on the > > same lines... > > The current answer is to use line isolators -- Radio Shack sells these > things as privacy widgets. Although I am certain that the SLIC the FXO > cards use has onhook/offhook detection, it's just a matter of seeing if you > can read that info and tell * that the line is in use. I've had the same issue as the original post, and have not found an acceptable way to handle it in an environment larger then a home office. The privacy widget approach isn't the real answer either since * blindly starts sending touchtone signaling with first listening for dial tone. Mark or anyone... is there any support within the X100P and associated software to implement a "listen for dialtone before dialing" approach? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie with questions
> One of the many areas that I am unclear on is if we can use > the existing phones from the old PBX system? Some are Meridian > M7310 and some are Norstar M7208 models. Nope. Those are proprietary digital keysets and will only work with Meridian/Norstar control units. > They each support multiple lines with a multi-line display, but > I haven't seen anything on this list about sporting multiple > lines on a single phone... There are phones out there that have more than one line appearance, like the Cisco 7940 or 7960 (with 2 and 6 lines respectively). One thing I'm personally not clear on is having what's commonly known as "multi line appearance", where phone A is extension 201, phone B is extension 202, and they both have an appearance of 222 which would appear/ring on both phones. The other type of multi line appearance is where 201 and 202 have appearances of each other. A call to 201 would only ring phone A, but phone B would have a light lit up to show that phone A is in use. This isn't an Asterisk specific thing, it goes back to the SIP standard/specification. It might be easier with MGCP, I'm not sure - I remember reading something about it but it was a while ago. Hope that helps. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wildcard TDM400P - FXO?
Does anyone know if or when the FXO daughter boards for the TDM400P will be available? Gene ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP
I wish you would take this stuff to personal email, I am tired of wasting my time reading this crap. If you idiots want to give "lesions" on how YOU would like people to post on list servers _DO_IT_VIA_PERSONAL_EMAIL_!!! None of the rest of us care. This is a personal messages from you to someone else. Stop wastering MY bandwidth. I personal like top posting as I don't have to scroll all the way to the bottom to read what is most of the time one damn sentence. So, it's not worth *your* time organizing your e-mail sensibly, but it's worth everyone else's time having to dig through lines of text to work out what the context is? I find that selfish, at best. Please see the following page (strong words warning). It pretty much sums it all up nicely: - http://thegestalt.org/simon/quoterant.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
OK OK OK, I got it. See my response inside the body of your E-mail. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Alastair Maw Sent: Tuesday, October 14, 2003 8:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) On 15/10/03 00:15, Uriel Carrasquilla wrote: > Does anybody else have a strong opinion one way or the other? If it > is left to John and myself we have a 1:1 vote. See how much easier it is to follow the thread of conversation if you quote just enough of the e-mail you're responding to so people know what's going on without having to read through pages of text? [URIEL] - I have to learn how to quote with Outlook. Please see RFC 1855: - http://www.faqs.org/rfcs/rfc1855.html Decent mail clients that behave sensibly regarding quoting are easy to come by. You can even set up Outlook to behave vaguely properly and quote using >. > As a matter of fact, I am of the opinion that the response to E-mails > should go at the top to save time. So, it's not worth *your* time organizing your e-mail sensibly, but it's worth everyone else's time having to dig through lines of text to work out what the context is? I find that selfish, at best. [URIEL] you are absolutely right and I do apologize. Ignorance is not an excuse. Please see the following page (strong words warning). It pretty much sums it all up nicely: - http://thegestalt.org/simon/quoterant.html [URIEL] Thank you. -- Al Maw ___ 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] No Ringback on Iconnect or Nikotel
When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no ringback tone when making a call. I have tried adding the ‘r’ command to the dial sting with no help. Does anyone else have this problem or offer any suggestions? Thanks, Kevin
Re: [Asterisk-Users] [OT] Proper quoting (was: NAT, SIP
learn.to can kiss my *** I'll top quote till death, And so will almost every other person on this planet. Tilghman Lesher wrote: On Tuesday 14 October 2003 18:15, Uriel Carrasquilla wrote: I have to tell you, at the expense of offending you, that I use MS-Outlook and the responses go to the tope of the messages. At work I use Lotus Notes and the same thing happens. Before, I used PROFS (on mainframes) and the same principle applied. All in all, 20+ years of using this principle for e-mails at both work and home. As a matter of fact, I am of the opinion that the response to E-mails should go at the top to save time. However, this is not about me but the * group and the well being of this list. Does anybody else have a strong opinion one way or the other? If it is left to John and myself we have a 1:1 vote. This is all you really need to know: http://learn.to/quote/ -Tilghman ___ 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] Newbie with questions
I have an existing Meridian PBX system that I am looking at replacing with Asterisk for the multi-office features. One of the many areas that I am unclear on is if we can use the existing phones from the old PBX system? Some are Meridian M7310 and some are Norstar M7208 models. They each support multiple lines with a multi-line display, but I haven't seen anything on this list about sporting multiple lines on a single phone... then again, there are years of message I have not read.
RE: [Asterisk-Users] [OT] Proper quoting (was: NAT, SIP (was: No sound with SIP Phones on the Internet))
Excellent points in the printed world. I am not certain that from mail to eMail I would use the same principles. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher Sent: Tuesday, October 14, 2003 6:53 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] [OT] Proper quoting (was: NAT, SIP (was: No sound with SIP Phones on the Internet)) On Tuesday 14 October 2003 18:15, Uriel Carrasquilla wrote: > I have to tell you, at the expense of offending you, that I use > MS-Outlook and the responses go to the tope of the messages. At work > I use Lotus Notes and the same thing happens. Before, I used PROFS > (on mainframes) and the same principle applied. All in all, 20+ > years of using this principle for e-mails at both work and home. As > a matter of fact, I am of the opinion that the response to E-mails > should go at the top to save time. However, this is not about me but > the * group and the well being of this list. Does anybody else have > a strong opinion one way or the other? If it is left to John and > myself we have a 1:1 vote. This is all you really need to know: http://learn.to/quote/ -Tilghman ___ 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] NAT, SIP (was: No sound with SIP Phones on the Internet)
On 15/10/03 00:15, Uriel Carrasquilla wrote: Does anybody else have a strong opinion one way or the other? If it is left to John and myself we have a 1:1 vote. See how much easier it is to follow the thread of conversation if you quote just enough of the e-mail you're responding to so people know what's going on without having to read through pages of text? Please see RFC 1855: - http://www.faqs.org/rfcs/rfc1855.html Decent mail clients that behave sensibly regarding quoting are easy to come by. You can even set up Outlook to behave vaguely properly and quote using >. As a matter of fact, I am of the opinion that the response to E-mails should go at the top to save time. So, it's not worth *your* time organizing your e-mail sensibly, but it's worth everyone else's time having to dig through lines of text to work out what the context is? I find that selfish, at best. Please see the following page (strong words warning). It pretty much sums it all up nicely: - http://thegestalt.org/simon/quoterant.html -- Al Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P to Adtran TA750 - No dialtone or ring
Don't forget to reverse the FXO/FXS in the TA750. They are opposite to the
asterisk config files.
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason Piterak
Sent: Tuesday, October 14, 2003 5:13 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] T100P to Adtran TA750 - No dialtone or ring
Hello all,
I've got a T100P connected to an Adtran TA750 with a T1 crossover...
This connects to a patch panel with phone ports. The Adtran is fully
populated with FXS cards.
All I get on any phone port is a fast clicking noise... No dialtone.
Asterisk 'sees' the card, (the channels show up in /proc/zaptel).
Incoming calls are routed to the zap/x channel, but no ring.
I'm hoping I'm overlooking something stupid.
Thanks ahead of time...
--Jason
Here are some (possibly) relevant snippits from various places:
o T100 LED shows green...
o Not showing any errors in /var/log/asterisk/messages (debug logging
enabled)
o Adtran config is set to:
--> 2. Provisioning Templates
--> 1. Factory Default (ESF,B8ZS,Loopstart)'
o OS/hardware:
System OS:
debian testing/unstable
kernel: Custom 2.4.22-ck
(Con Kolivas patch set:
o Preempt
o Low-latency
o AA vm hacks
o RL2 Desktop Tuning
o Debian logo in FB)
Digium Cards:
T100P -->FXS
X100P -->FXO
o Asterisk version:
Asterisk CVS-10/02/03-17:52:20
--
asterisk:~# cat /proc/zaptel/1
Span 1: WCT1/0 "Digium Wildcard T100P T1/PRI Card 0" B8ZS/ESF
IRQ misses: 21904
1 WCT1/0/1 FXOKS (In use)
2 WCT1/0/2 FXOKS (In use)
3 WCT1/0/3 FXOKS (In use)
4 WCT1/0/4 FXOKS (In use)
5 WCT1/0/5 FXOKS (In use)
6 WCT1/0/6 FXOKS (In use)
7 WCT1/0/7 FXOKS (In use)
8 WCT1/0/8 FXOKS (In use)
9 WCT1/0/9 FXOKS (In use)
10 WCT1/0/10 FXOKS (In use)
11 WCT1/0/11 FXOKS (In use)
12 WCT1/0/12 FXOKS (In use)
13 WCT1/0/13 FXOKS (In use)
14 WCT1/0/14 FXOKS (In use)
15 WCT1/0/15 FXOKS (In use)
16 WCT1/0/16 FXOKS (In use)
17 WCT1/0/17 FXOKS (In use)
18 WCT1/0/18 FXOKS (In use)
19 WCT1/0/19 FXOKS (In use)
20 WCT1/0/20 FXOKS (In use)
21 WCT1/0/21 FXOKS (In use)
22 WCT1/0/22 FXOKS (In use)
23 WCT1/0/23 FXOKS (In use)
24 WCT1/0/24 FXOKS (In use)
asterisk:~# cat /proc/zaptel/2
Span 2: WCFXO/0 "Wildcard X101P Board 1"
25 WCFXO/0/0 FXSKS (In use)
---
asterisk:/etc/asterisk# cat /etc/zaptel.conf
#T1:
span=1,0,0,esf,b8zs
fxoks=1-24
loadzone = us
defaultzone=us
#X100P - Single-line FXO card
fxsks=25
-
asterisk:/etc/asterisk# cat zapata.conf
...
[channels]
;T1-fxo (incomming channels) on the channel bank
;-
;
;context = bell
;language = en
;signalling = fxs_ks
;usecallerid = yes
;hidecallerid = no
;echocancel = yes
;echocancelwhenbridged = no
;;if immediate is set to yes, asterisk will automatically answer the line
;;and jump to the 's' extension for the context.
;;immediate = yes
;group = 1
;channel => 1
;T1-fxs (inside handsets) on the channel bank
context = local
language = en
signalling = fxo_ks
rxwink = 300
usecallerid = yes
hidecallerid = no
callwaiting = no
;callwaitingcallerid=yes <--Change if Callwaiting is yes
threewaycalling = yes
transfer = yes
cancelforward = yes
callreturn = no
echocancel = yes
echocanelwhenbridged = no
immediate = no
rxgain=0.0
txgain=0.0
channel => 1-24
...
;SinglePort-fxo (incomming channels)
context = bell
language = en
signalling = fxs_ks
usecallerid = yes
hidecallerid = no
echocancel = yes
echocancelwhenbridged = no
;if immediate is set to yes, asterisk will automatically answer the line
;and jump to the 's' extension for the context.
immediate = yes
group = 1
channel => 25
-
Console during a call being routed to zap/1:
-- Starting simple switch on 'Zap/25-1'
-- Executing BackGround("Zap/25-1", "thankyou") in new stack
-- Playing 'thankyou'
== CDR updated on Zap/25-1
-- Executing Goto("Zap/25-1", "mainmenu|s|2") in new stack
-- Goto (mainmenu,s,2)
-- Executing BackGround("Zap/25-1", "greeting-announcements") in new
stack
-- Playing 'greeting-announcements'
== CDR updated on Zap/25-1
-- Executing Goto("Zap/25-1", "routing|300|1") in new stack
-- Goto (routing,300,1)
-- Executing Macro("Zap/25-1", "oneline|300|Zap/1") in new stack
-- Executing DBget("Zap/25-1", "fwdexten=CFU/300") in new stack
-- DBget: varname=fwdexten, family=CFU, key=300
-- DBget: Value not found in database.
-- Executing Goto("Zap/25-1", "s|4") in new stack
-- Goto (macro-oneline,s,4)
-- Executing Dial("Zap/25-1", "Zap/1||") in new stack
-- Called 1
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Hungup 'Zap/1-1'
== Spawn extension (macro-oneline, s, 4) exited non-zero on 'Zap/25-1' in
macro 'oneline'
== Spawn extension (routing, s, 1) exited non-zero on 'Zap/25-1'
asterisk:~# lsmod
Module Size Used byNot tainted
so
Re: [Asterisk-Users] Digium cards just for timing - moden cards?
I think the X101P uses an Ambient/Intel HAM 56K modem chip. The older X100 is based on Motorola. Any authoritative answers? Andrew Kohlsmith wrote: Is that why there is an X100P and an X101P? What design is the X101P based on? AFAIK the current design uses a Tiger 320 chip which is essentially a PCI gateway -- it provides a serial port and an 8-bit parallel interface to anything. The single FXO card uses the serial interface, and the single T1 card uses the 8-bit parallel interface to a standard Dallas Semiconductor T1/E1/J1 framer IC. Regards, Andrew ___ 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] */SER/FW
Steve: Unless Asterisk is on the public side of the Internet, you will run into problems if the UA (SIP phones) are behind a NAT. In the scenario you presented, I think SER would be used for all calls between SIP phones and they would only go to Asterisk when you need to Gateway into the PSTN some of the calls. Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 6:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] */SER/FW Hi, I've just read the postings regarding the interworking between * and SER. As these persons seem quite knowledgeable on this, I would like to have their advise on my planned installation: - I have broadband cable access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan to install Asteriks on that server - I plan to install a SIP-proxy,registrar on the same server (I've been looking at iptel's SER) - I plan to use the Budgetone SIP phones - I plan to have a public (static) IP address All this to have my own little phone company for me and my family/friends as we are spread over Europe (high international phone costs!). Calling eachother on our SIP phones and also being able to use eachother's PBX's to make local calls. I would host the SIP Registrar (in stead of outsourcing). My main question lies in the interworking between iptel's SER and Asteriks. Not only on the configuration side, but also on the network side (here I mean: can both run on the same server, or do they need to have different IP addresses, ...). Does anybody have a diagram or any practical guide on how to install this? Thanks, Steven ___ 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] Digium cards just for timing - moden cards?
On Tue, 14 Oct 2003 19:06:42 -0400, Jon Pounder <[EMAIL PROTECTED]> wrote: At 06:56 PM 10/14/2003, you wrote: Two comments: 1) "just a $10 winmodem?" is this literally true? then what $10 card is known to work? Have you tried one? Yes it is pretty clear that a Winmodem card could work for this application yes its literally true. The cards are out of production now afaik, but were based on a motorola dsp. (And yes they were available for under $10USD, and tested and worked fine) Does a list exist that has all the winmodems that can supply timing? All I would like to do is get rid of the echo in meetme. 2) I've always though a real DSP based modem card could be re-programmed to a much more interesting use. Those on-board TI DSP chips are quite powerfull computers. Easly enough to compute any audio codec or even the front end for speech understanding. they are flash reprogramable too. The lindsl opensource project is attempting to do this, and reprogram a winmodem to do much higher "DSL like" speeds in a point to point twisted pair network than a conventional modem would do. Last I knew interest by the author had fell off though. > I don't think an X100P card will help. Anything you gain from the > ztdummy driver will be the same as what you can gain from an X100P, > FWIW > the card is just a $10 winmodem. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.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 -- Using M2, Opera's revolutionary e-mail client: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Chris: What I think makes the SER solution attractive is the fact that Asterisk tends to drop registration of SIP phones. I agree, SER works like a Gatekeeper and for that matter IAX when the two end-points can be bridged. Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Albertson Sent: Tuesday, October 14, 2003 2:36 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) --- Uriel Carrasquilla <[EMAIL PROTECTED]> wrote: > John: > are you aware of any documentation on how to configre SER to be a > front-end > to Asterisk? > I suspect it is very inexpensive to put a SER server in a hosting > facility I think the cost is about the same as for putting a web server at a hosting facility. But I don't think you need high bandwidth. SER simply sets up the call. I don't think the audio data actually goes through SER. It goes directly between the two end points. This is the big problem with using Asterisk for SIP. With Asterisk the audio data between two SIP extensions has to actualy go into then out of the Asterisk box. This does not scale well to thousands of users like in a university campus or a comercial SIP service. > to forward traffic to multiple Asterisks based on Least Cost Routing. > My problem is that my experience is with Asterisk and not with SER. > Uriel > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of John Todd > Sent: Monday, October 13, 2003 8:11 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones > on > the Internet) > > > >I'm curently looking into using SER to front end SIP calls for > >Asterisk. > >Basicaly all SIP users would register with SER not Asterisk and then > >Asterisk and SER exchange registrations. > > > >SER is a very capable SIP router, much more sophisticated than > Asterisk > >as it can look inside packets and route based on what it finds or > even > >re-write packets based on user specified logic. > > > >SER is GPL'd and has very good user documentation. Don't know how > well > >the above will work. The claim by the authors or SER that it can > >handle thousands of calls per second is quite impressive > > > >One other nice feature is that SER users can set up their own SIP > >accounts using a web interface and not needing to edit *.conf > files. > > > >See here for details http://www.iptel.org/ser/ > > > > > >= > >Chris Albertson > > Home: 310-376-1029 [EMAIL PROTECTED] > > Cell: 310-990-7550 > > Office: 310-336-5189 [EMAIL PROTECTED] > > KG6OMK > > SER is an excellent option as a front end to Asterisk. It is a > "true" SIP proxy, whereas Asterisk is a hybrid, and SIP has not been > the primary focus of Asterisk development. In fact, Asterisk's SIP > implementation is very limited (though it is extremely pragmatic.) > > However, moving to SER does not solve any of the issues about the > proxy being behind a NAT, and I believe that SER will have the same > problems (though I could be wrong on this; I haven't experimented > with SER's ability to work from behind a NAT.) SIP clients work > well enough behind NAT (most of them, anyway) but the servers are a > different story. > > I really like SER's third-party addons for account administration; > Asterisk is significantly more complex, and probably would not be as > easily converted to such a front end. In fact, SER has a very > complex routing/scripting language that is not easily administered > with a web front end, so I think that SER and Asterisk suffer from > the same problems. If someone were to come up with a simple way to > administer voicemail.conf and sip.conf from a web tool, that would go > far to making Asterisk a bit more user-accessible... > > JT > ___ > 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.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] Line in use detection...
> I would like to know if is possible to setup my Asterisk to detect if the > phone lines from FXO cards are in use. We use the parallel phones on the > same lines... The current answer is to use line isolators -- Radio Shack sells these things as privacy widgets. Although I am certain that the SLIC the FXO cards use has onhook/offhook detection, it's just a matter of seeing if you can read that info and tell * that the line is in use. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium cards just for timing - moden cards?
> 2) I've always though a real DSP based modem card could be >re-programmed to a much more interesting use. Those on-board >TI DSP chips are quite powerfull computers. Easly enough to >compute any audio codec or even the front end for speech >understanding. they are flash reprogramable too. That and without blasting the PC with interrupts... I wonder if the codec conversion costs would be better (not sure how well SSE fares against a good DSP) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Andre:
This makes a lot of sense. I had used Asterisk in the past to play the role
of Gatekeeper for directing traffic to the appropriate Asterisk acting as a
PSTN gateway. IAX does a heck of a good job in that configuration.
However, with SIP, I have run into nothing but trouble with registrations
falling off.
I have read the SER manual I am going to jump into it, now that I know that
in "practice" it works and it is not only theory in a manual.
Thank you,
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andres
Sent: Tuesday, October 14, 2003 12:43 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)
On Monday 13 October 2003 22:26, Uriel Carrasquilla wrote:
> John:
> are you aware of any documentation on how to configre SER to be a
front-end
> to Asterisk?
Hi Uriel,
At TeleSIP we run a cluster of several geographically distributed SER
Servers
that hande all our SIP Routing. SER is a robust, fast and stable platform
which has worked flawlessly for us. We use * as our company PBX and PSTN
Gateway. Basically what you need to do is to device a numbering plan so
that
SERs routing logic can forward the call to * when it needs to.
For example in ser.cfg you could put something like this:
#
###PSTN ACCESS###
#
if (method=="INVITE") {
if (uri=~"sip:[EMAIL PROTECTED]") {
log(1, "This is a Long Distance Call\n");
route(6);
break;
};
};
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P Echo Problems..What's going to happen?
John: I don't use MSN so I can't comment. I do know that when my connections are pure VoIP (no analog PSTN connections), the quality is better if enough bandwidth is available. TCP is a protocol that gets used when you want to make sure a packet arrives at the other end. UDP is better for voice because you don't want packets to be retransmitted and have to wait to assemble them on the other end in sequence so the conversation makes sense. Regards, Uriel -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of John MSent: Monday, October 13, 2003 11:18 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P Echo Problems..What's going to happen? What I don’t understand is why MSN messenger is perfect with no echo? I switch back and forth and still hear a big difference. I believe MSN is using TCP rather than UDP. Can * run on TCP rather than UDP? I think this makes senses and can eliminate that echo. -Original Message-From: Uriel Carrasquilla [mailto:[EMAIL PROTECTED] Sent: Monday, October 13, 2003 10:02 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P Echo Problems..What's going to happen? John: I have been around voice over data packets for quite a few years and I am still to see the perfect system that works identical to circuit switching 100% of the time. My opinion is that there is a lot more to the story than just parameters. Packet loses, double compressions, faulty routers, bandwidth, analog to digital and so on can get in the way. On the other hand, if your customer understand the benefits, and I mean more than cost, and can leave with 80% perfect, then you will be able to understand why a lot of companies have opted for VoIP (or ATM or Frame Relay). Regards, Uriel -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of John MSent: Monday, October 13, 2003 1:41 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] X100P Echo Problems..What's going to happen?Importance: High I’ve read and experienced the echo problems with the X100P. Is Digium going to fix the problem or refund our money? I want to see this work because myself and other small companies out there use analog lines. I would trade up to T1 but that requires me to have at least 9 lines. If I did trade up, do the T1 cards work perfectly with no echo at all? I get echo with my directly connected computer using Xten SIP. No matter with all the suggestions to change the parameters, it still has echo. Does anyone have the T1 and have no problems at all? I would surely appreciate you experiences. What’s my option to get this too work flawlessly? John
RE: [Asterisk-Users] Mitel 5055 phone
-Original Message- From: Paul Crick [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 2:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Mitel 5055 phone I'd like to find out more about it too.. Are the buttons programmable as line appearances and speed dials too? Does anyone have one working as a standalone SIP phone with no Mitel PBX/other hardware in the mix? I've worked extensively on SX2000's and have to say I love those systems (anyone need any work doing on one?!) Cheers Paul [messages re-ordered] I am working with one now. As far as I have gotten is that I can authenticate against asterisk but then can't make a call. I will be spending more time on this this week. Too many projects. I don't know what price you're looking at but Mitel just dropped the price of these phones to be pretty much the same as their regular Minet IP phones so they may be even cheaper than you think. Also early next year Mitel will be releasing new software for their 5220 phone which will allow it to dual boot between Minet and SIP. Barry Barry - Please post your summary to the list when you are completed. Another decent higher-end phone is good news for everyone, and if you can find out the specifics of why the Mitel doesn't work with *, perhaps the rest of the community can come up with some patch ideas. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium cards just for timing - moden cards?
> Is that why there is an X100P and an X101P? What design is the X101P > based on? AFAIK the current design uses a Tiger 320 chip which is essentially a PCI gateway -- it provides a serial port and an 8-bit parallel interface to anything. The single FXO card uses the serial interface, and the single T1 card uses the 8-bit parallel interface to a standard Dallas Semiconductor T1/E1/J1 framer IC. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P setup in Switzerland
Marcel:
Some switches with particular functionalities don't expect ton=unknown. For
example, Lucent 5ESS crap (in EuroISDN) doesn't work pretty well using
unknown.
In Siemens EWSD switch, if you have a divided PRI in two differents
Directory Numbers, you cannot send unknown.
In fact, send unknown is more comfortable, I know, but is not the better
resolution, neither A number nor B number.
Regards,
Gus
- Original Message -
From: "Marcel Prisi" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, October 14, 2003 7:32 PM
Subject: Re: [Asterisk-Users] E100P setup in Switzerland
> Marcel Prisi a écrit :
>
> > I have some news ... after a bit of tweaking, the following seems to
> > work with Swisscom :
> >
> > span=1,1,0,ccs,hdb3,crc4
> > bchan=1-15,17-31
> > dchan=16
> >
> > I still have a problem : Incoming calls from PRI work well, but
> > outgoing call don't :
> >
> > I dial from a Grandstream 101
> >
> > -- Executing Dial("SIP/22-9129", "Zap/g1/234234234") in new stack
> > -- Called g1/234234234
> > -- Zap/1-1 is ringing
> >
> > Well ... I hear a ringing tone in the Grandstream, but the called
> > number doesn't ring at all ...
> >
> > What can I do ??
> >
> > Thanks
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> Funny to reply to myself ... just for the sake of it, simply added
>
> pridialplan=unknown
>
> in zapata.conf before the channels specifications and it works
> perfectly well !
>
> And thanks to Digium technical support !
>
> ___
> 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] H.323 - SIP gateway
You shouldn't treat asterisk as a gatekeeper (because it ain't) On your H.323 equipment, set asterisk up as a gateway. > Hi! > > I have done that but it doesn't work because I need also the port 1720 to make > the comunication. Port 1719 is only used to the RAS messages and 1720 is used > to > make the communication. > > Thanks a lot for your help > > Regards, > > Mireia > Quoting CW_ASN - Gus <[EMAIL PROTECTED]>: > > > Or, if you must use 1719, try to change h323.conf: > > > > [general] > > port = 1719 > > bindaddr = 0.0.0.0 > > > > Regards, > > > > Gus > > > > - Original Message - > > From: "Eric Wieling" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Tuesday, October 14, 2003 12:23 PM > > Subject: Re: [Asterisk-Users] H.323 - SIP gateway > > > > > > > h323 runs on port 1720. Your gatekeeper is trying to contact the wrong > > > port number. > > > > > > On Tue, 2003-10-14 at 10:02, Mireia Munoz de jesus wrote: > > > > Hi all! > > > > > > > > Please I need someone that have already done an H.323 - SIP gateway to > > help me > > > > with some problems. I can stablish calls from a SIP telephone to a > > H.323, but I > > > > can't do vice versa... (problems with port 1719- when the gatekeeper > > tries to > > > > contact with asterisk at this port, it is unrecheable...). > > > > > > > > Please someone can help me? > > > > > > > > Regards, > > > > > > > > Mireia > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > > > Sample configs and more: http://www.fnords.org/~eric/asterisk/ > > > > > > BTEL Consulting > > > +1-850-484-4535 x2111 (Pensacola) > > > +1-504-595-3916 x2111 (New Orleans) > > > +1-877-677-9643 x2111 (Toll Free) > > > > > > ___ > > > 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] Digium cards just for timing - moden cards?
Is that why there is an X100P and an X101P? What design is the X101P based on? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jon Pounder > Sent: Tuesday, October 14, 2003 7:07 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Digium cards just for timing - moden cards? > > pamAssassin 2.55 (1.174.2.19-2003-05-19-exp) > > At 06:56 PM 10/14/2003, you wrote: > > >Two comments: > > > >1) "just a $10 winmodem?" is this literally true? then what > >$10 card is known to work? Have you tried one? Yes it is pretty > >clear that a Winmodem card could work for this application > > yes its literally true. The cards are out of production now afaik, but > were > based on a motorola dsp. (And yes they were available for under $10USD, > and > tested and worked fine) > > > >2) I've always though a real DSP based modem card could be > >re-programmed to a much more interesting use. Those on-board > >TI DSP chips are quite powerfull computers. Easly enough to > >compute any audio codec or even the front end for speech > >understanding. they are flash reprogramable too. > > The lindsl opensource project is attempting to do this, and reprogram a > winmodem to do much higher "DSL like" speeds in a point to point twisted > pair network than a conventional modem would do. > > Last I knew interest by the author had fell off though. > > > > > > > I don't think an X100P card will help. Anything you gain from the > > > ztdummy driver will be the same as what you can gain from an X100P, > > > FWIW > > > the card is just a $10 winmodem. > > > > > >= > >Chris Albertson > > Home: 310-376-1029 [EMAIL PROTECTED] > > Cell: 310-990-7550 > > Office: 310-336-5189 [EMAIL PROTECTED] > > KG6OMK > > > >__ > >Do you Yahoo!? > >The New Yahoo! Shopping - with improved product search > >http://shopping.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 BEGIN:VCARD VERSION:2.1 N:Joakimsen;Andrew FN:Andrew Joakimsen ([EMAIL PROTECTED]) ORG:Envision Studio TEL;WORK;VOICE:(888) 210-8063 TEL;CELL;VOICE:(305) 776-0334 TEL;WORK;FAX:(305) 669-6720 EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20030819T050904Z END:VCARD
[Asterisk-Users] re: Restoring Cisco 7960 to defaults
Can anyone point me to some online documentation showing how to reset a CP-7960 to factory default settings. I have some that are configured for Callmanager and I want to get them back to generic default config. Any info is appreciated. Thanks Cory Andrews ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon Diva Server BRI (T1) Cards
For T-1 use a Digium card for about US$500 On Tue, 2003-10-14 at 17:02, Roger Schreiter wrote: > Hi, > > my asterisk experiences with isdn cards supported by i4l > are not very good, but with avm a1 and capi everything > works very fine and stable. (SuSE Linx 8.2, Kernel 2.4.20, > german ISDN). > > Now I want to connect a T1. Should I use an AVM T1-B > for approx 6000 EUR or is it ok to use one of Eicon's cheaper > Diva Server BRI S2M cards? > > Do they use i4l or is there a similar thing as AVM's capi > driver for eicon's S2M cards? > > Are there any experiences how they compare when used > with asterisk? > > > Thanks for any reports! > > > Roger. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone Tone
Yes, of course. However, that would be a feature of the SIP phone, not Asterisk, since Asterisk isn't providing the dialtone on your SIP phone, the phone is doing that. On Tue, 2003-10-14 at 16:28, Chris Hariga wrote: > Hi, > > si posible on SIP phones to have the dial tone after 9 like on the FXS card? > I set ignorepat => 9 on my extensions.conf... > > Best regards, > > Chris HARIGA > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards just for timing - moden cards?
At 06:56 PM 10/14/2003, you wrote: Two comments: 1) "just a $10 winmodem?" is this literally true? then what $10 card is known to work? Have you tried one? Yes it is pretty clear that a Winmodem card could work for this application yes its literally true. The cards are out of production now afaik, but were based on a motorola dsp. (And yes they were available for under $10USD, and tested and worked fine) 2) I've always though a real DSP based modem card could be re-programmed to a much more interesting use. Those on-board TI DSP chips are quite powerfull computers. Easly enough to compute any audio codec or even the front end for speech understanding. they are flash reprogramable too. The lindsl opensource project is attempting to do this, and reprogram a winmodem to do much higher "DSL like" speeds in a point to point twisted pair network than a conventional modem would do. Last I knew interest by the author had fell off though. > I don't think an X100P card will help. Anything you gain from the > ztdummy driver will be the same as what you can gain from an X100P, > FWIW > the card is just a $10 winmodem. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.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
[Asterisk-Users] Dial-out using Asterisk
I have to dial out to a customer site ... We now only have our E1 & Asterisk to call out. I think I can set-up an ATA-186 and a regular analog modem to do it, but is it possible to use on of the 32 channels of the E1 as a 64kbit/s PPP ISDN dial-out line ? How ?? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards just for timing - moden cards?
Two comments: 1) "just a $10 winmodem?" is this literally true? then what $10 card is known to work? Have you tried one? Yes it is pretty clear that a Winmodem card could work for this application 2) I've always though a real DSP based modem card could be re-programmed to a much more interesting use. Those on-board TI DSP chips are quite powerfull computers. Easly enough to compute any audio codec or even the front end for speech understanding. they are flash reprogramable too. > I don't think an X100P card will help. Anything you gain from the > ztdummy driver will be the same as what you can gain from an X100P, > FWIW > the card is just a $10 winmodem. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Proper quoting (was: NAT, SIP (was: No sound with SIP Phones on the Internet))
On Tuesday 14 October 2003 18:15, Uriel Carrasquilla wrote: > I have to tell you, at the expense of offending you, that I use > MS-Outlook and the responses go to the tope of the messages. At work > I use Lotus Notes and the same thing happens. Before, I used PROFS > (on mainframes) and the same principle applied. All in all, 20+ > years of using this principle for e-mails at both work and home. As > a matter of fact, I am of the opinion that the response to E-mails > should go at the top to save time. However, this is not about me but > the * group and the well being of this list. Does anybody else have > a strong opinion one way or the other? If it is left to John and > myself we have a 1:1 vote. This is all you really need to know: http://learn.to/quote/ -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P to Adtran TA750 - No dialtone or ring
Jason Piterak wrote: Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm hoping I'm overlooking something stupid. Thanks ahead of time... --Jason Does the light on front of the TA750 show that the channel is up? Can you do diags thru the TA750 admin port to your phones? ring, etc.? Do you have the most current TA750 firmware (L35?)? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P setup in Switzerland
Marcel Prisi a écrit :
I have some news ... after a bit of tweaking, the following seems to
work with Swisscom :
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
I still have a problem : Incoming calls from PRI work well, but
outgoing call don't :
I dial from a Grandstream 101
-- Executing Dial("SIP/22-9129", "Zap/g1/234234234") in new stack
-- Called g1/234234234
-- Zap/1-1 is ringing
Well ... I hear a ringing tone in the Grandstream, but the called
number doesn't ring at all ...
What can I do ??
Thanks
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Funny to reply to myself ... just for the sake of it, simply added
pridialplan=unknown
in zapata.conf before the channels specifications and it works
perfectly well !
And thanks to Digium technical support !
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iaxtel and Voicepulse
I found the answer: One can disallow a codec within each context of iax.conf. Stig - Original Message - From: Stig Hess To: [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 11:43 PM Subject: Re: [Asterisk-Users] Iaxtel and Voicepulse Voicepulse _has_ ilbc turned on, but it will only work if I disallow GSM. So I wondered if there was some way to turn on the codecs for every connection... Stig - Original Message - From: Brian West To: [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 11:08 PM Subject: Re: [Asterisk-Users] Iaxtel and Voicepulse You must use GSM with iaxtel and Voicepulse for now... I talked to the guyfrom voicepulse and they said ilbc might be turned sometime in the future.But not sure.bkwOn Tue, 14 Oct 2003, Stig Hess wrote:> I'm having trouble configuring these services the way I want. Basically I> prefer using iLBC before GSM, however Iaxtel only want to talk GSM. It> _seems_ that Voicepulse prefers GSM also, because even if I put ILBC before> GSM in the "allow=" part of iax.conf, if GSM is mentioned, Voicepulse will> use it. If I don't allow GSM Voicepulse will use ILBC.>> Does anyone know how to achieve this? Can "allow=" be put into each> extension/provider definition?>> Thanks,>> __> ºªº Stig Hess ºªº> Iaxtel: 1-700-2854373> USA: +1-212-400-7920> Norge: +47-51665681>___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Uriel - 1) Please stop top-posting. 2) I'm afraid I don't have any data on specifics of creating a front-end. I know how to do it, but my time these days is spent writing lots of other projects that I have been doing. :-) I would suggest you get SER and set it up - it's quite easy, and the documentation on SER itself is very well written, and if you have a good idea of how SIP works you should be able to patch together an appropriate system. However, if you aren't 100% familiar with how SIP works, I would stick to just an Asterisk system; SER doesn't allow for any of the "shortcuts" that Asterisk has. 3) Use Google and do some searching. I found some quick links with a few of the keywords that would seem obvious, but I don't have enough time to review them... JT John: Thank you for responding. I am in the process of installing SER and hope to have it ready by this weekend. I am in the process of installing some equipment at a local colo. I have to tell you, at the expense of offending you, that I use MS-Outlook and the responses go to the tope of the messages. At work I use Lotus Notes and the same thing happens. Before, I used PROFS (on mainframes) and the same principle applied. All in all, 20+ years of using this principle for e-mails at both work and home. As a matter of fact, I am of the opinion that the response to E-mails should go at the top to save time. However, this is not about me but the * group and the well being of this list. Does anybody else have a strong opinion one way or the other? If it is left to John and myself we have a 1:1 vote. Regards, Uriel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line in use detection...
Hi, I would like to know if is possible to setup my Asterisk to detect if the phone lines from FXO cards are in use. We use the parallel phones on the same lines... Best regards, Chris HARIGA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pattern matching problem when dialing
I am having problems with early dialing and chan_phone. In extensions.conf I have: exten => _41.,1,Dial,IAX If I dial via a SIP or ZAP channels it works fine. With chan_phone it start dialing right after the 3rd number. If tried different combinations like (41., ... or _41X., ) and still the same problem. This used to work ok a few weeks back.!!
Re: [Asterisk-Users] Eicon Diva Server BRI (T1) Cards
On Wed, 15 Oct 2003, Roger Schreiter wrote: > Now I want to connect a T1. Should I use an AVM T1-B > for approx 6000 EUR or is it ok to use one of Eicon's cheaper > Diva Server BRI S2M cards? Why not use Digium hardware like the Wildcard T100P or Wildcard TE410P? Regards, Jac -- Jac KersingTechnical Consultant The-Box Development [EMAIL PROTECTED] http://www.the-box.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco hard IP phones and Skinny vs. SIP
I have Asterisk up and running and it is working great with my SIP phones. However, I have some "Skinny"-protocol Cisco 7960s. Does Asterisk support the Skinny protocol? I've seen some references to Skinny in the software. If so, should I stick with Skinny with the 7960 or convert to SIP? If anyone has some Skinny confs they would send me I'd be much obliged. If I should convert to SIP... how? I've seen the Cisco directions but they involve downloading software from Cisco that I can't download because I don't have a maintenance contract. Is there a way around this, or am I better off sticking with Skinny? Thanks! Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXTEL - Problem Configuration.
Somebody keeps saying it is bad form to respond to one's own postings, but
I am going to do it here ...
Further experimentation and I discovered that this change to
iax.conf make the problem go away:
;
; Trust Caller*ID Coming from iaxtel.com
;
;[iaxtel]
;type=friend
;context=default
;auth=rsa
;inkeys=iaxtel
;[iaxtel2]
;
; Backwards compatible entry for IAXtel pre-RSA
;
type=friend
context=default
deny=0.0.0.0/0.0.0.0
permit=12.37.165.130/255.255.255.255
Michael
On Tue, 14 Oct 2003, Michael T Farnworth wrote:
> Is your problem related to the settings under iax.conf? I have commented
> out the whole of [iaxtel2] and left in [iaxtel].
>
> I have a problem with my setup in that I have got it to register by
> putting the register command in the iax.conf, but when I call my own
> number I just get silence. Output on console is:
>
> -- Executing Dial("SIP/phone1-b4ee",
> "IAX/farnwomt:[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack
> -- Calling using options
> 'exten=17008451426;callerid=192.168.254.160;language=en;context=iaxtel;username=farnwomt;formats=2;capability=65283;version=1;adsicpe=2'
> -- Called farnwomt:[EMAIL PROTECTED]/[EMAIL PROTECTED]
> -- Call accepted by 12.37.165.130 (format GSM)
> -- Format for call is GSM
>
> Any guesses why this happens?
>
> Thanks,
> Michael
>
> On Tue, 14 Oct 2003, Ariel Batista wrote:
>
> > Ok folks I have another question. So far I have gotten my IAXTEL number
> > and I have been able to make calls from my asterisk system to any IAXTEL
> > number and even to FWD numbers. I also got FWD to work and I now can get
> > calls to my main system. It's great when these things work. But when I
> > call my own IAXTEL number 17005441100 all I get is a message saying the
> > user is un registered or un available. I set the register =>
> > loginName:[EMAIL PROTECTED] in the iax.conf. So how do I configure
> > the extensions.conf to send the call to my extension! Here is where I
> > am lost!
>
> > ___
> > 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] Digium cards just for timing
Martin, Thanks a lot. The problem was a turned on silence suppression on cisco ata 186. Now it seems to work perfectly. Thanks to everybody else too. Michael On Tuesday 14 October 2003 05:04 pm, Martin Pycko wrote: > With the musiconhold and SIP-SIP call it turnes out that you need to > disable silence supporesion on your phones/gateways since the timing is > taken from the coming stream (but only for musiconhold AFAIK) > > regards > Martin > > On Tue, 14 Oct 2003, Michael Ulitskiy wrote: > > > Hi, > > > > I've found that neither Michael Manousos patch nor ztdummy driver > > do not fix musiconhold sound interruption problem up to acceptable quality > > level. Sound is choppy here anyway. > > It is my understanding (please correct me if I'm wrong) that if I have > > a Digium card in my asterisk machine, these problems should be gone > > 'cause those cards provide some reliable timing. So I have no choice > > and wish to buy a cheapest Digium card just for timing. I have no PSTN > > ports, it's pure voip environment here. > > So my question is whether any Digium card would be ok or I have to buy > > some specific card? I'm looking at X100P card as it is the cheapest one. > > Would it be enough? > > Thank you. > > > > Michael > > > > > > ___ > > 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] Eicon Diva Server BRI (T1) Cards
Hi, my asterisk experiences with isdn cards supported by i4l are not very good, but with avm a1 and capi everything works very fine and stable. (SuSE Linx 8.2, Kernel 2.4.20, german ISDN). Now I want to connect a T1. Should I use an AVM T1-B for approx 6000 EUR or is it ok to use one of Eicon's cheaper Diva Server BRI S2M cards? Do they use i4l or is there a similar thing as AVM's capi driver for eicon's S2M cards? Are there any experiences how they compare when used with asterisk? Thanks for any reports! Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Certified Hardware label?
Concept sounds good, not sure whether this was intentional or not but the logo just says "WiFi Certified" if I don't give it more than a quick glance. Something like this should be distinguishable without actually reading it from all other similar certification or conformance marks. Eg: how many certifications are there for dvd players, and we can all tell just by shape which one it is without actually looking closely. Simply a large asterisk of a specific color and that's it would be much more identifyable to those who care. At 05:43 PM 10/14/2003, you wrote: Hello, While I've been searching for SIP hardphones and trying to pick between all of the different features that are available for them I had the idea that we (The Asterisk community) should create a label that hard-phone manufacturers can put on their products that meet basic requirements to function with the Asterisk VOIP system. Or, at the very least, a listing on a website that would show which features work on each specific hard-phone. You would think this would be as simple as saying "It says SIP compatible so it must run with Asterisk", but as I found out when I had a new Teledex SIP phone to test for a week and never got it to register with Asterisk, that is not necessarily true. I would be willing to register a domain and design and host the website, as well as be a contact for manufacturers and a point-man for Asterisk Users. I'm sure this would probably need the approval of the guys at Digium, but I think it would greatly help the community in general as well as promote Asterisk in the telecom industry. We may even be able to charge manufacturers for the "privalege" to be Asterisk Certified or sell them stickers and have all the money go to Digium for development. http://www.freedomphones.net/asterisk_certified.gif Let me know what you think. MATT--- ___ 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] dialling out
No I am using a Cisco 7940 Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Besch Sent: Tuesday, 14 October 2003 11:13 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dialling out Mick, If you're using the Grandstream, there appears to be a bug in early dial. After the 4th or 5th digit, * sends back an address incomplete message and the phone responds with the buzy - generated at the phone. A SIP debug trace indicates that * is sending the correct information and continues to waut for the next digit, which never arrives. I have sent Grandstream a bug report and they are working on the problem. As a temporary workaround turn the early dial option off. If you are not using the Grandstream, then there may be a similar problem in other phones, or, despite appearances from the SIP debug trace, there may be a problem with *; I'm betting on the phone in this instance. Turn on SIP debug in the CLI and study the responses after each digit. Ignoring the acknowlege packets, you should get a series of invites from the phone, each subsequent one with one additional digit, and each followed by an address incomplete response from *. If everything is working, this happens until the invite requests an extension that is actually in the dialplan. Stephen R. Besch [EMAIL PROTECTED] wrote: >When trying to dial out > >982420173 our main number > >I get the engaged signal before I finish entering the phone number > >Any ideas > > > > >Regards Mick > >___ >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] Digium cards just for timing
- Hi, I've found that neither Michael Manousos patch nor ztdummy driver do not fix musiconhold sound interruption problem up to acceptable quality level. Sound is choppy here anyway. It is my understanding (please correct me if I'm wrong) that if I have a Digium card in my asterisk machine, these problems should be gone 'cause those cards provide some reliable timing. So I have no choice and wish to buy a cheapest Digium card just for timing. I have no PSTN ports, it's pure voip environment here. So my question is whether any Digium card would be ok or I have to buy some specific card? I'm looking at X100P card as it is the cheapest one. Would it be enough? Thank you. Michael - Michael, Have you tried zaprtc from http://www.junghanns.net/asterisk ? It seems to be working here on a 'SIP only' configuration without any problems. Music on hold sounds perfect. -Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXTEL - Problem Configuration.
Is your problem related to the settings under iax.conf? I have commented
out the whole of [iaxtel2] and left in [iaxtel].
I have a problem with my setup in that I have got it to register by
putting the register command in the iax.conf, but when I call my own
number I just get silence. Output on console is:
-- Executing Dial("SIP/phone1-b4ee",
"IAX/farnwomt:[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack
-- Calling using options
'exten=17008451426;callerid=192.168.254.160;language=en;context=iaxtel;username=farnwomt;formats=2;capability=65283;version=1;adsicpe=2'
-- Called farnwomt:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 12.37.165.130 (format GSM)
-- Format for call is GSM
Any guesses why this happens?
Thanks,
Michael
On Tue, 14 Oct 2003, Ariel Batista wrote:
> Ok folks I have another question. So far I have gotten my IAXTEL number
> and I have been able to make calls from my asterisk system to any IAXTEL
> number and even to FWD numbers. I also got FWD to work and I now can get
> calls to my main system. It's great when these things work. But when I
> call my own IAXTEL number 17005441100 all I get is a message saying the
> user is un registered or un available. I set the register =>
> loginName:[EMAIL PROTECTED] in the iax.conf. So how do I configure
> the extensions.conf to send the call to my extension! Here is where I
> am lost!
> ___
> 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] Iaxtel and Voicepulse
Voicepulse _has_ ilbc turned on, but it will only work if I disallow GSM. So I wondered if there was some way to turn on the codecs for every connection... Stig - Original Message - From: Brian West To: [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 11:08 PM Subject: Re: [Asterisk-Users] Iaxtel and Voicepulse You must use GSM with iaxtel and Voicepulse for now... I talked to the guyfrom voicepulse and they said ilbc might be turned sometime in the future.But not sure.bkwOn Tue, 14 Oct 2003, Stig Hess wrote:> I'm having trouble configuring these services the way I want. Basically I> prefer using iLBC before GSM, however Iaxtel only want to talk GSM. It> _seems_ that Voicepulse prefers GSM also, because even if I put ILBC before> GSM in the "allow=" part of iax.conf, if GSM is mentioned, Voicepulse will> use it. If I don't allow GSM Voicepulse will use ILBC.>> Does anyone know how to achieve this? Can "allow=" be put into each> extension/provider definition?>> Thanks,>> __> ºªº Stig Hess ºªº> Iaxtel: 1-700-2854373> USA: +1-212-400-7920> Norge: +47-51665681>___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mitel 5055 phone
Where did you get your 5055? I've tried to order it from 2 separate vendors and they both tell me it will be at least a month before they get any in. MATT--- -Original Message- From: Barry Porch [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 4:48 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Mitel 5055 phone I am working with one now. As far as I have gotten is that I can authenticate against asterisk but then can't make a call. I will be spending more time on this this week. Too many projects. I don't know what price you're looking at but Mitel just dropped the price of these phones to be pretty much the same as their regular Minet IP phones so they may be even cheaper than you think. Also early next year Mitel will be releasing new software for their 5220 phone which will allow it to dual boot between Minet and SIP. Barry -Original Message- From: Paul Crick [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 2:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Mitel 5055 phone I'd like to find out more about it too.. Are the buttons programmable as line appearances and speed dials too? Does anyone have one working as a standalone SIP phone with no Mitel PBX/other hardware in the mix? I've worked extensively on SX2000's and have to say I love those systems (anyone need any work doing on one?!) Cheers Paul ___ 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] Asterisk Certified Hardware label?
Hello, While I've been searching for SIP hardphones and trying to pick between all of the different features that are available for them I had the idea that we (The Asterisk community) should create a label that hard-phone manufacturers can put on their products that meet basic requirements to function with the Asterisk VOIP system. Or, at the very least, a listing on a website that would show which features work on each specific hard-phone. You would think this would be as simple as saying "It says SIP compatible so it must run with Asterisk", but as I found out when I had a new Teledex SIP phone to test for a week and never got it to register with Asterisk, that is not necessarily true. I would be willing to register a domain and design and host the website, as well as be a contact for manufacturers and a point-man for Asterisk Users. I'm sure this would probably need the approval of the guys at Digium, but I think it would greatly help the community in general as well as promote Asterisk in the telecom industry. We may even be able to charge manufacturers for the "privalege" to be Asterisk Certified or sell them stickers and have all the money go to Digium for development. http://www.freedomphones.net/asterisk_certified.gif Let me know what you think. MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Phone Tone
Hi, si posible on SIP phones to have the dial tone after 9 like on the FXS card? I set ignorepat => 9 on my extensions.conf... Best regards, Chris HARIGA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P to Adtran TA750 - No dialtone or ring
Hello all,
I've got a T100P connected to an Adtran TA750 with a T1 crossover...
This connects to a patch panel with phone ports. The Adtran is fully
populated with FXS cards.
All I get on any phone port is a fast clicking noise... No dialtone.
Asterisk 'sees' the card, (the channels show up in /proc/zaptel).
Incoming calls are routed to the zap/x channel, but no ring.
I'm hoping I'm overlooking something stupid.
Thanks ahead of time...
--Jason
Here are some (possibly) relevant snippits from various places:
o T100 LED shows green...
o Not showing any errors in /var/log/asterisk/messages (debug logging enabled)
o Adtran config is set to:
--> 2. Provisioning Templates
--> 1. Factory Default (ESF,B8ZS,Loopstart)'
o OS/hardware:
System OS:
debian testing/unstable
kernel: Custom 2.4.22-ck
(Con Kolivas patch set:
o Preempt
o Low-latency
o AA vm hacks
o RL2 Desktop Tuning
o Debian logo in FB)
Digium Cards:
T100P -->FXS
X100P -->FXO
o Asterisk version:
Asterisk CVS-10/02/03-17:52:20
--
asterisk:~# cat /proc/zaptel/1
Span 1: WCT1/0 "Digium Wildcard T100P T1/PRI Card 0" B8ZS/ESF
IRQ misses: 21904
1 WCT1/0/1 FXOKS (In use)
2 WCT1/0/2 FXOKS (In use)
3 WCT1/0/3 FXOKS (In use)
4 WCT1/0/4 FXOKS (In use)
5 WCT1/0/5 FXOKS (In use)
6 WCT1/0/6 FXOKS (In use)
7 WCT1/0/7 FXOKS (In use)
8 WCT1/0/8 FXOKS (In use)
9 WCT1/0/9 FXOKS (In use)
10 WCT1/0/10 FXOKS (In use)
11 WCT1/0/11 FXOKS (In use)
12 WCT1/0/12 FXOKS (In use)
13 WCT1/0/13 FXOKS (In use)
14 WCT1/0/14 FXOKS (In use)
15 WCT1/0/15 FXOKS (In use)
16 WCT1/0/16 FXOKS (In use)
17 WCT1/0/17 FXOKS (In use)
18 WCT1/0/18 FXOKS (In use)
19 WCT1/0/19 FXOKS (In use)
20 WCT1/0/20 FXOKS (In use)
21 WCT1/0/21 FXOKS (In use)
22 WCT1/0/22 FXOKS (In use)
23 WCT1/0/23 FXOKS (In use)
24 WCT1/0/24 FXOKS (In use)
asterisk:~# cat /proc/zaptel/2
Span 2: WCFXO/0 "Wildcard X101P Board 1"
25 WCFXO/0/0 FXSKS (In use)
---
asterisk:/etc/asterisk# cat /etc/zaptel.conf
#T1:
span=1,0,0,esf,b8zs
fxoks=1-24
loadzone = us
defaultzone=us
#X100P - Single-line FXO card
fxsks=25
-
asterisk:/etc/asterisk# cat zapata.conf
...
[channels]
;T1-fxo (incomming channels) on the channel bank
;-
;
;context = bell
;language = en
;signalling = fxs_ks
;usecallerid = yes
;hidecallerid = no
;echocancel = yes
;echocancelwhenbridged = no
;;if immediate is set to yes, asterisk will automatically answer the line
;;and jump to the 's' extension for the context.
;;immediate = yes
;group = 1
;channel => 1
;T1-fxs (inside handsets) on the channel bank
context = local
language = en
signalling = fxo_ks
rxwink = 300
usecallerid = yes
hidecallerid = no
callwaiting = no
;callwaitingcallerid=yes <--Change if Callwaiting is yes
threewaycalling = yes
transfer = yes
cancelforward = yes
callreturn = no
echocancel = yes
echocanelwhenbridged = no
immediate = no
rxgain=0.0
txgain=0.0
channel => 1-24
...
;SinglePort-fxo (incomming channels)
context = bell
language = en
signalling = fxs_ks
usecallerid = yes
hidecallerid = no
echocancel = yes
echocancelwhenbridged = no
;if immediate is set to yes, asterisk will automatically answer the line
;and jump to the 's' extension for the context.
immediate = yes
group = 1
channel => 25
-
Console during a call being routed to zap/1:
-- Starting simple switch on 'Zap/25-1'
-- Executing BackGround("Zap/25-1", "thankyou") in new stack
-- Playing 'thankyou'
== CDR updated on Zap/25-1
-- Executing Goto("Zap/25-1", "mainmenu|s|2") in new stack
-- Goto (mainmenu,s,2)
-- Executing BackGround("Zap/25-1", "greeting-announcements") in new stack
-- Playing 'greeting-announcements'
== CDR updated on Zap/25-1
-- Executing Goto("Zap/25-1", "routing|300|1") in new stack
-- Goto (routing,300,1)
-- Executing Macro("Zap/25-1", "oneline|300|Zap/1") in new stack
-- Executing DBget("Zap/25-1", "fwdexten=CFU/300") in new stack
-- DBget: varname=fwdexten, family=CFU, key=300
-- DBget: Value not found in database.
-- Executing Goto("Zap/25-1", "s|4") in new stack
-- Goto (macro-oneline,s,4)
-- Executing Dial("Zap/25-1", "Zap/1||") in new stack
-- Called 1
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Hungup 'Zap/1-1'
== Spawn extension (macro-oneline, s, 4) exited non-zero on 'Zap/25-1' in macro
'oneline'
== Spawn extension (routing, s, 1) exited non-zero on 'Zap/25-1'
asterisk:~# lsmod
Module Size Used byNot tainted
soundcore 3876 0 (autoclean)
wcusb 21888 0 (unused)
wcfxo 8352 1
wct1xxp12704 24
zaptel180064 54 [wcusb wcfxo wct1xxp]
ppp_generic17668 0 [zaptel]
slhc5216 0 [ppp_generic]
serial 48228 1 (auto
[Asterisk-Users] 200-400ms latency
Has anyone tested using SIP endpoints (Possibly the ATA-186) with a connection that has at least 200ms, if not more, of latency? We are trying to get some stuff setup in Australia and wanted to know if this would be feasable, are there any added delays? Echos?
Re: [Asterisk-Users] Iaxtel and Voicepulse
You must use GSM with iaxtel and Voicepulse for now... I talked to the guy from voicepulse and they said ilbc might be turned sometime in the future. But not sure. bkw On Tue, 14 Oct 2003, Stig Hess wrote: > I'm having trouble configuring these services the way I want. Basically I > prefer using iLBC before GSM, however Iaxtel only want to talk GSM. It > _seems_ that Voicepulse prefers GSM also, because even if I put ILBC before > GSM in the "allow=" part of iax.conf, if GSM is mentioned, Voicepulse will > use it. If I don't allow GSM Voicepulse will use ILBC. > > Does anyone know how to achieve this? Can "allow=" be put into each > extension/provider definition? > > Thanks, > > __ > ºªº Stig Hess ºªº > Iaxtel: 1-700-2854373 > USA: +1-212-400-7920 > Norge: +47-51665681 > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTEL - Problem Configuration.
Ok folks I have another question. So far I have gotten my IAXTEL number and I have been able to make calls from my asterisk system to any IAXTEL number and even to FWD numbers. I also got FWD to work and I now can get calls to my main system. It's great when these things work. But when I call my own IAXTEL number 17005441100 all I get is a message saying the user is un registered or un available. I set the register => loginName:[EMAIL PROTECTED] in the iax.conf. So how do I configure the extensions.conf to send the call to my extension! Here is where I am lost! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards just for timing
No. I also run machines with pure VoIP and there is not a single problem with music on hold. I don't think an X100P card will help. Anything you gain from the ztdummy driver will be the same as what you can gain from an X100P, FWIW the card is just a $10 winmodem. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Michael Ulitskiy > Sent: Tuesday, October 14, 2003 4:46 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Digium cards just for timing > > Hi, > > I've found that neither Michael Manousos patch nor ztdummy driver > do not fix musiconhold sound interruption problem up to acceptable quality > level. Sound is choppy here anyway. > It is my understanding (please correct me if I'm wrong) that if I have > a Digium card in my asterisk machine, these problems should be gone > 'cause those cards provide some reliable timing. So I have no choice > and wish to buy a cheapest Digium card just for timing. I have no PSTN > ports, it's pure voip environment here. > So my question is whether any Digium card would be ok or I have to buy > some specific card? I'm looking at X100P card as it is the cheapest one. > Would it be enough? > Thank you. > > Michael > > > ___ > 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] Turning a regular call into a conference?
What steps would have to happen, in order to take an already-connected call and move both parties into a conference room? i.e. do both parties have to be parked first, or can one or both of them just be immediately transferred to a MeetMe extension? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium cards just for timing
With the musiconhold and SIP-SIP call it turnes out that you need to disable silence supporesion on your phones/gateways since the timing is taken from the coming stream (but only for musiconhold AFAIK) regards Martin On Tue, 14 Oct 2003, Michael Ulitskiy wrote: > Hi, > > I've found that neither Michael Manousos patch nor ztdummy driver > do not fix musiconhold sound interruption problem up to acceptable quality > level. Sound is choppy here anyway. > It is my understanding (please correct me if I'm wrong) that if I have > a Digium card in my asterisk machine, these problems should be gone > 'cause those cards provide some reliable timing. So I have no choice > and wish to buy a cheapest Digium card just for timing. I have no PSTN > ports, it's pure voip environment here. > So my question is whether any Digium card would be ok or I have to buy > some specific card? I'm looking at X100P card as it is the cheapest one. > Would it be enough? > Thank you. > > Michael > > > ___ > 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] H.323 - SIP gateway
Hi! I have done that but it doesn't work because I need also the port 1720 to make the comunication. Port 1719 is only used to the RAS messages and 1720 is used to make the communication. Thanks a lot for your help Regards, Mireia Quoting CW_ASN - Gus <[EMAIL PROTECTED]>: > Or, if you must use 1719, try to change h323.conf: > > [general] > port = 1719 > bindaddr = 0.0.0.0 > > Regards, > > Gus > > - Original Message - > From: "Eric Wieling" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, October 14, 2003 12:23 PM > Subject: Re: [Asterisk-Users] H.323 - SIP gateway > > > > h323 runs on port 1720. Your gatekeeper is trying to contact the wrong > > port number. > > > > On Tue, 2003-10-14 at 10:02, Mireia Munoz de jesus wrote: > > > Hi all! > > > > > > Please I need someone that have already done an H.323 - SIP gateway to > help me > > > with some problems. I can stablish calls from a SIP telephone to a > H.323, but I > > > can't do vice versa... (problems with port 1719- when the gatekeeper > tries to > > > contact with asterisk at this port, it is unrecheable...). > > > > > > Please someone can help me? > > > > > > Regards, > > > > > > Mireia > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > > Sample configs and more: http://www.fnords.org/~eric/asterisk/ > > > > BTEL Consulting > > +1-850-484-4535 x2111 (Pensacola) > > +1-504-595-3916 x2111 (New Orleans) > > +1-877-677-9643 x2111 (Toll Free) > > > > ___ > > 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] Digium cards just for timing
Hi, I've found that neither Michael Manousos patch nor ztdummy driver do not fix musiconhold sound interruption problem up to acceptable quality level. Sound is choppy here anyway. It is my understanding (please correct me if I'm wrong) that if I have a Digium card in my asterisk machine, these problems should be gone 'cause those cards provide some reliable timing. So I have no choice and wish to buy a cheapest Digium card just for timing. I have no PSTN ports, it's pure voip environment here. So my question is whether any Digium card would be ok or I have to buy some specific card? I'm looking at X100P card as it is the cheapest one. Would it be enough? Thank you. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mitel 5055 phone
I am working with one now. As far as I have gotten is that I can authenticate against asterisk but then can't make a call. I will be spending more time on this this week. Too many projects. I don't know what price you're looking at but Mitel just dropped the price of these phones to be pretty much the same as their regular Minet IP phones so they may be even cheaper than you think. Also early next year Mitel will be releasing new software for their 5220 phone which will allow it to dual boot between Minet and SIP. Barry -Original Message- From: Paul Crick [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 2:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Mitel 5055 phone I'd like to find out more about it too.. Are the buttons programmable as line appearances and speed dials too? Does anyone have one working as a standalone SIP phone with no Mitel PBX/other hardware in the mix? I've worked extensively on SX2000's and have to say I love those systems (anyone need any work doing on one?!) Cheers Paul ___ 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] Iaxtel and Voicepulse
I'm having trouble configuring these services the way I want. Basically I prefer using iLBC before GSM, however Iaxtel only want to talk GSM. It _seems_ that Voicepulse prefers GSM also, because even if I put ILBC before GSM in the "allow=" part of iax.conf, if GSM is mentioned, Voicepulse will use it. If I don't allow GSM Voicepulse will use ILBC. Does anyone know how to achieve this? Can "allow=" be put into each extension/provider definition? Thanks, __ºªº Stig Hess ºªº Iaxtel: 1-700-2854373USA: +1-212-400-7920Norge: +47-51665681
[Asterisk-Users] Setting Outbound Caller ID for T1 link
Can someone Check something out for me here. I have a PBX behind an
asterisk system connnected via T1. The PBX is not seeing the caller ID or
ANI coming across from asterisk. I am setting it explicitly using :
extensions.conf fragment
[macro-dialswitch3]
; ARG1 Called Number, Arg2 Caller ID NUM, Arg3 Caller ID
exten => s,1,SetCallerID(${ARG3},a)
exten => s,2,SetAccount(AP${ARG1})
exten => s,3,Macro(startmonitor,${ARG1},${ARG2},${CHANNEL},ib)
exten => s,4,NoOp(${CALLERID})
exten => s,5,Dial(Zap/g4/${ARG1})
exten => s,6,Congestion
exten => s,106,Busy
I
The asterisk box should be acting as the PSTN to the internal switch here.
Am I missing something.
zapta.conf fragment
;This is the t1 to switch3 used for outbound calls
signalling = em_w
group=4
context=outbound
callerid=asreceived
channel => 73-96
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] list server Delays
Is it common on this list to experience long posting delays?
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
On Tue, Oct 14, 2003 at 04:20:27PM -0300, CW_ASN - Gus wrote: > Walker: > > "sip show channel" refers to a Call ID: > > noc2pbx2*CLI> sip show channels > Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter > Format > 172.16.254.620341522910 3607139911@ 00101/3 0ms ms ALAW > 1 active SIP channel(s) > > Then, you could see the details: > > noc2pbx2*CLI> sip show channel [EMAIL PROTECTED] > Call-ID: [EMAIL PROTECTED] > Our Codec Capability: 524302 > Non-Codec Capability: 1 > Joint Codec Capability: 12 > Theoretical Address: 172.16.254.62:5060 > Received Address:172.16.254.62:5060 > NAT Support: No > Our Tag: 1906405977 > Their Tag: 1054824328 > Need Destroy:0 > DTMF Mode: rfc2833 > > > Hope this helps, > > Gus Great, Gus! Also to Martin. I put 2 and 2 together and figured it out. The tab key does work when you use the correct channel. My system has a longer "Call ID" than your example, Gus. Just for documentation, here's what it looks like on my version: asterisk*CLI> sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.206206 79010fe1-5a 00101/61141 0ms ms ULAW ... 8 active SIP channel(s) asterisk*CLI> sip show channel [EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] ... I typed in the Call ID and hit tab, voila! Thanks, Walker > > > - Original Message - > From: "Walker Haddock" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, October 14, 2003 3:37 PM > Subject: [Asterisk-Users] use of SIP SHOW CHANNELS question > > > > I am trying to figure out the correct syntax for the CLI command "SIP SHOW > CHANNELS". I have tried > > SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is > connected such as: > > > > -- Zap/15-1 is ringing > > -- Zap/15-1 answered SIP/206-4299 > > asterisk*CLI> sip show channel SIP/206-4299 > > No such SIP Call ID 'SIP/206-4299' > > > > > > I always get the "No such SIP Call ID ..." > > > > Thanks, Walker > > -- > > DataCrest, Inc. -- Technically Superior ** > > Walker Haddock http://www.datacrest.com > > DataCrest, Inc.e-mail: [EMAIL PROTECTED] > > 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 > > Birmingham, AL 35216 fax: 1-205-823-7838 > > *** > > ___ > > 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 -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI function calls
I am guessing the AGI command "SET EXTENSION " allow for you to set which extension the rest of the call will occur over? So a call comes into the switch and I could make the AGI script check the DID or DNIS which is really in the variable agi_dnid? If not how can I get the DID and what is agi_dnid for? After that I can do a database lookup from the script and then issue the "SET EXTENSION" command to asterisk to allocate the call to the right extension if available? I went to http://home.cogeco.ca /~camstuff/agi.html For the AGI command descriptions. Is there a better/more up to date site? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ACD/IVR dialogs/SIP/client environment
So does “SET EXTENSION ” allow for you to set which extension the rest of the call will occur over? So a call comes into the switch and I could make the AGI script check the DID or DNIS which is really in the variable agi_dnid? After that I can do a database lookup from the script and then issue the “SET EXTENSION” command to asterisk to allocate the call to the right extension if available? I had to go to http://home.cogeco.ca /~camstuff/agi.html For the AGI command descriptions. Is there a better/more up to date site? -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent: Monday, October 13, 2003 3:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ACD/IVR dialogs/SIP/client environment Have you looked into using AGI? You can use any language you like (C++, PERL, PHP, even bash shell scripts) But I would look into the Asterisk::AGI perl module: http://asterisk.gnuinter.net/ because it can already interface with Asterisk and pass the variables onto your script. I actually haven’t found much information regarding AGI in other languages, so this would be your best bet. It should be very easy to implement if you already have the application that can accept input with the callers number and do its stuff. Regards, Andrew Joakimsen Envision Studio http://envisionstudio.net 888-210-8063 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nate Clifford Sent: Monday, October 13, 2003 4:11 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ACD/IVR dialogs/SIP/client environment Ok I have tried to post to this list server but have just gotten the automated reply saying the moderator has to approve it to the list first which was my mistake for sending from the wrong email account. So if the moderator finally approves my questions and you see the same post again “Sorry”. My situation is this: I havn't installed Asterisk yet but am curious the general way you would go about doing an ACD to A SIP client and causing an application to pop on the client side. I had thought that the way to do that would be to trigger two different events from inside an IVR on the Asterisk server. Is that possible? Can you send the call into a dialog/code that will determine the client via the DID/DNIS and then call and pass variable to an application that will communicate over the network with the clients app and then load a web page. Then the next part of the dialog/code would initiate the SIP session with the clients station? Any answers or replies to help distill this question would help. Thanks.
Re: [Asterisk-Users] Outgoing CallerID
JanM wrote:
> Hello,
>
> Does anyone know how to set the outgoing CallerID properly when using
> Snom200/SIP/CAPI/BRI?
>
> Following doesn´t work:
> exten => _0.,1,SetCallerID,526910
> exten => _0.,2,Dial,CAPI/526980:${EXTEN:1}
>
> Asterisk writes:
> *CLI> -- Executing SetCallerID("SIP/226-ada0", "526910") in new stack
> -- Executing Dial("SIP/226-ada0", "CAPI/526980:0408665390") in new stack
> -- Called 526980:0408665390
> -- CAPI[contr3/526980]/0 is ringing
>
> My mobile is only showing some other number that my isdn line is having.
>
> ---JanM---
Telecom restrictions?
I can set only caller IDs within the set of numbers provided me from telecom.
Check with your telecom if you're allowed to set any caller ID.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
On Tue, Oct 14, 2003 at 01:52:14PM -0500, Martin Pycko wrote: > Use tabulator button for asterisk to help you guess the name. I have been trying that. I think that may have been implemented after I did the CSV. I'm at: Asterisk CVS-09/24/03-20:51:12 built by [EMAIL PROTECTED] on a i686 running Linux I try typing: sip show channel SIP/[Tab][Tab][Tab] like with bash and nothing ever pops up. What I really want to do is figure out what the channel name is so I can use in in the Channel Status command in the AGI. > > regards > Martin > > On Tue, 14 Oct 2003, Walker Haddock wrote: > > > I am trying to figure out the correct syntax for the CLI command "SIP SHOW > > CHANNELS". I have tried > > SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is connected > > such as: > > > > -- Zap/15-1 is ringing > > -- Zap/15-1 answered SIP/206-4299 > > asterisk*CLI> sip show channel SIP/206-4299 > > No such SIP Call ID 'SIP/206-4299' > > > > > > I always get the "No such SIP Call ID ..." > > > > Thanks, Walker > > -- > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] On an RH9 box, where does wcusb get loaded?
If you do "make config" in the zaptel then it's going to be loaded during bootup. Otherwise it's not being loaded unless you do 'modprobe wcusb' regards Martin On 14 Oct 2003, tom wrote: > >From - > Received: from rwcrmhc12.comcast.net ([216.148.227.85]) by > sccrmxc11.comcast.net (sccrmxc11) with ESMTP id > <20031014185753s1100nos46e>; Tue, 14 Oct 2003 18:57:53 + > Received: from rwcrmhc12.comcast.net (localhost[127.0.0.1]) by > comcast.net > (rwcrmhc12) with ESMTP id <20031014185753014006qmtie>; Tue, 14 Oct 2003 > 18:57:53 + > From: Mail Delivery Subsystem <[EMAIL PROTECTED]> > Subject: Returned mail: delivery problems encountered > Message-Id: <[EMAIL PROTECTED]@[EMAIL PROTECTED]> > Date: 14 Oct 2003 18:57:51 + > To: <[EMAIL PROTECTED]> > Mime-Version: 1.0 > Content-Type: multipart/report; report-type=delivery-status; > boundary="_3f8c472f.7355.0+comcast.net=_" > X-Evolution-Source: pop://[EMAIL PROTECTED]/ > > > --_3f8c472f.7355.0+comcast.net=_ > Content-Type: text/plain > > A message (from <[EMAIL PROTECTED]>) was received at 14 Oct 2003 > 18:57:41 +. > > The following addresses had delivery problems: > > <[EMAIL PROTECTED]> > Permanent Failure: > 550_5.1.1_<[EMAIL PROTECTED]>..._User_unknown > Delivery last attempted at Tue, 14 Oct 2003 18:57:50 - > > --_3f8c472f.7355.0+comcast.net=_ > Content-Type: message/delivery-status > > Reporting-MTA: dns; comcast.net > Arrival-Date: 14 Oct 2003 18:57:41 + > > Final-Recipient: rfc822; <[EMAIL PROTECTED]> > Action: failed > Status: 5.0.0 550_5.1.1_<[EMAIL PROTECTED]>..._User_unknown > Diagnostic-Code: smtp; Permanent Failure: Other undefined Status > Last-Attempt-Date: Tue, 14 Oct 2003 18:57:50 - > > --_3f8c472f.7355.0+comcast.net=_ > Content-Type: message/rfc822 > > Received: from [192.168.0.33] > (dnvr-dsl-gw28-poolc85.dnvr.uswest.net[65.101.254.85]) by comcast.net > (rwcrmhc12) with SMTP id <20031014185741014008t5g9e> (Authid: > landslide_x); > Tue, 14 Oct 2003 18:57:42 + > Subject: On RH9, where is wcusb loaded? > From: tom <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Content-Type: text/plain > Organization: > Message-Id: <[EMAIL PROTECTED]> > Mime-Version: 1.0 > X-Mailer: Ximian Evolution 1.2.2 (1.2.2-4) > Date: 14 Oct 2003 12:58:57 -0600 > Content-Transfer-Encoding: 7bit > > > I have a dev kit lite, and I'd like to have asterisk up and running when > I boot my linux box, but there are couple of things that are preventing > this from happening. First and foremost, the wcusb and zaptel modules > are loaded at startup, but wcxfo is not. In order to get everything > running (and loaded in the correct order), I have to remove wcusb, load > wcfxo, reload wcusb, and then run ztcfg. Any ideas on how I might all of > this loading in the correct order so that I won't have to keeps putzing > with it when I boot. > > Thanks. > > Tom > > > --_3f8c472f.7355.0+comcast.net=_-- > > > > ___ > 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] use of SIP SHOW CHANNELS question
Walker: "sip show channel" refers to a Call ID: noc2pbx2*CLI> sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 172.16.254.620341522910 3607139911@ 00101/3 0ms ms ALAW 1 active SIP channel(s) Then, you could see the details: noc2pbx2*CLI> sip show channel [EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Our Codec Capability: 524302 Non-Codec Capability: 1 Joint Codec Capability: 12 Theoretical Address: 172.16.254.62:5060 Received Address:172.16.254.62:5060 NAT Support: No Our Tag: 1906405977 Their Tag: 1054824328 Need Destroy:0 DTMF Mode: rfc2833 Hope this helps, Gus - Original Message - From: "Walker Haddock" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, October 14, 2003 3:37 PM Subject: [Asterisk-Users] use of SIP SHOW CHANNELS question > I am trying to figure out the correct syntax for the CLI command "SIP SHOW CHANNELS". I have tried > SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is connected such as: > > -- Zap/15-1 is ringing > -- Zap/15-1 answered SIP/206-4299 > asterisk*CLI> sip show channel SIP/206-4299 > No such SIP Call ID 'SIP/206-4299' > > > I always get the "No such SIP Call ID ..." > > Thanks, Walker > -- > DataCrest, Inc. -- Technically Superior ** > Walker Haddock http://www.datacrest.com > DataCrest, Inc.e-mail: [EMAIL PROTECTED] > 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 > Birmingham, AL 35216 fax: 1-205-823-7838 > *** > ___ > 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] On an RH9 box, where does wcusb get loaded?
>From - Received: from rwcrmhc12.comcast.net ([216.148.227.85]) by sccrmxc11.comcast.net (sccrmxc11) with ESMTP id <20031014185753s1100nos46e>; Tue, 14 Oct 2003 18:57:53 + Received: from rwcrmhc12.comcast.net (localhost[127.0.0.1]) by comcast.net (rwcrmhc12) with ESMTP id <20031014185753014006qmtie>; Tue, 14 Oct 2003 18:57:53 + From: Mail Delivery Subsystem <[EMAIL PROTECTED]> Subject: Returned mail: delivery problems encountered Message-Id: <[EMAIL PROTECTED]@[EMAIL PROTECTED]> Date: 14 Oct 2003 18:57:51 + To: <[EMAIL PROTECTED]> Mime-Version: 1.0 Content-Type: multipart/report; report-type=delivery-status; boundary="_3f8c472f.7355.0+comcast.net=_" X-Evolution-Source: pop://[EMAIL PROTECTED]/ --_3f8c472f.7355.0+comcast.net=_ Content-Type: text/plain A message (from <[EMAIL PROTECTED]>) was received at 14 Oct 2003 18:57:41 +. The following addresses had delivery problems: <[EMAIL PROTECTED]> Permanent Failure: 550_5.1.1_<[EMAIL PROTECTED]>..._User_unknown Delivery last attempted at Tue, 14 Oct 2003 18:57:50 - --_3f8c472f.7355.0+comcast.net=_ Content-Type: message/delivery-status Reporting-MTA: dns; comcast.net Arrival-Date: 14 Oct 2003 18:57:41 + Final-Recipient: rfc822; <[EMAIL PROTECTED]> Action: failed Status: 5.0.0 550_5.1.1_<[EMAIL PROTECTED]>..._User_unknown Diagnostic-Code: smtp; Permanent Failure: Other undefined Status Last-Attempt-Date: Tue, 14 Oct 2003 18:57:50 - --_3f8c472f.7355.0+comcast.net=_ Content-Type: message/rfc822 Received: from [192.168.0.33] (dnvr-dsl-gw28-poolc85.dnvr.uswest.net[65.101.254.85]) by comcast.net (rwcrmhc12) with SMTP id <20031014185741014008t5g9e> (Authid: landslide_x); Tue, 14 Oct 2003 18:57:42 + Subject: On RH9, where is wcusb loaded? From: tom <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Content-Type: text/plain Organization: Message-Id: <[EMAIL PROTECTED]> Mime-Version: 1.0 X-Mailer: Ximian Evolution 1.2.2 (1.2.2-4) Date: 14 Oct 2003 12:58:57 -0600 Content-Transfer-Encoding: 7bit I have a dev kit lite, and I'd like to have asterisk up and running when I boot my linux box, but there are couple of things that are preventing this from happening. First and foremost, the wcusb and zaptel modules are loaded at startup, but wcxfo is not. In order to get everything running (and loaded in the correct order), I have to remove wcusb, load wcfxo, reload wcusb, and then run ztcfg. Any ideas on how I might all of this loading in the correct order so that I won't have to keeps putzing with it when I boot. Thanks. Tom --_3f8c472f.7355.0+comcast.net=_-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help - Cisco 7960 re-certification
Can someone please point me to a Cisco reseller who can re-certify a 7960 an put it under a maintenance agreement? Paul Mahler [EMAIL PROTECTED] phone: 650-207-9855 fax: 877-408-0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ariel Batista Sent: Tuesday, October 14, 2003 10:40 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] "Gates steps up telecom campaign" -- Original Message -- From: Steven Critchfield <[EMAIL PROTECTED]> >> I understand I used to do the same. Sometimes up to 2 times a week. >> But Asterisk has problems with Zombie lines. And guess what you have >> to do to get them unlocked? > >If you are getting them that often, you may want to sit near the user >who occupies that line and see what they do odd that causes the zombies. >It would help with the debuging. > >We only seem them every couple of months here. > I have and found it due to the Nortel 452 and the 390 phones being so slow. It mainly happens on our receptions transfering calls to our support queue. It start thinking it need to setup a 3 way call! Then it will connect the transfer call correctly. About 20 secounds later it thinks the 3rd call is valid then sends it back to the original extension and that is when it goes out! I have grown to hate these phones the 452 and 390. >-- >Steven Critchfield <[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
Re: [Asterisk-Users] managers.conf Clarification Question
On Tuesday 14 October 2003 13:24, Anthony Minessale wrote: > Does Anyone have a breakdown on what each option means in > manager.conf > > system,call,log,verbose,command,agent,user > > I want to make a user who does not get a ton of events in > the socket and is just for sending a query and getting that 1 reply > > I dont want to keep restarting my pbx to figure it out. > > I'm sure some may be self-explanatory but I was wondering if anyone > knows for sure which options are which. The states are all defined in include/asterisk/manager.h: #define EVENT_FLAG_SYSTEM (1 << 0) /* System events such as module load/unload */ #define EVENT_FLAG_CALL (1 << 1) /* Call event, such as state change, etc */ #define EVENT_FLAG_LOG (1 << 2) /* Log events */ #define EVENT_FLAG_VERBOSE (1 << 3) /* Verbose messages */ #define EVENT_FLAG_COMMAND (1 << 4) /* Ability to read/set commands */ #define EVENT_FLAG_AGENT(1 << 5) /* Ability to read/set agent info */ #define EVENT_FLAG_USER (1 << 6) /* Ability to read/set user info */ -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
Use tabulator button for asterisk to help you guess the name. regards Martin On Tue, 14 Oct 2003, Walker Haddock wrote: > I am trying to figure out the correct syntax for the CLI command "SIP SHOW > CHANNELS". I have tried > SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is connected > such as: > > -- Zap/15-1 is ringing > -- Zap/15-1 answered SIP/206-4299 > asterisk*CLI> sip show channel SIP/206-4299 > No such SIP Call ID 'SIP/206-4299' > > > I always get the "No such SIP Call ID ..." > > Thanks, Walker > -- > DataCrest, Inc. -- Technically Superior ** > Walker Haddock http://www.datacrest.com > DataCrest, Inc.e-mail: [EMAIL PROTECTED] > 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 > Birmingham, AL 35216 fax: 1-205-823-7838 > *** > ___ > 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] Mitel 5055 phone
I'd like to find out more about it too.. Are the buttons programmable as line appearances and speed dials too? Does anyone have one working as a standalone SIP phone with no Mitel PBX/other hardware in the mix? I've worked extensively on SX2000's and have to say I love those systems (anyone need any work doing on one?!) Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] use of SIP SHOW CHANNELS question
I am trying to figure out the correct syntax for the CLI command "SIP SHOW CHANNELS". I have tried SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is connected such as: -- Zap/15-1 is ringing -- Zap/15-1 answered SIP/206-4299 asterisk*CLI> sip show channel SIP/206-4299 No such SIP Call ID 'SIP/206-4299' I always get the "No such SIP Call ID ..." Thanks, Walker -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] */SER/FW
I have asterisk and SER running on the same machine perfectly. As far as I am concerned the best way to do it is to have two ip addresses for the same ethernet interface. That way you can bind asterisk to one IP address and SER to the other. This way you don't have to use non-standard ports for your sip proxies. Good luck Jason > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Chris Albertson > Sent: 14 October 2003 07:21 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] */SER/FW > > > > I think the way to run SER and * on the same box is to have > Asterisk listen to SIP only on the loopback interface > 127.0.0.1 that way no SIP clients will ever connect to Asterisk. > > Configure SER to use all the interfaces. SER will connect to > SIP clients over the external interface and will notice > Asterisk on 127.0.0.1 > > Configure SER to route SIP to SIP calls and forward the rest > to Asterisk. The details depend on what you are trying to do > and maybe even the caller IDs of the SIP clients? It just depends. > > In my case I'd like any SIP user in the world to be able to > call me just like any PSTN user in the world can call me but > I also would like to provide a service to freinds and family > that would let them make SIP calls to each other, no need to > involve Asterisk in that. I don't want the audio data going > over my DSL link. > > The reason one might want to use SER is because it can do > things with SIP that Asterisk can't. SER and Asterisk don't > compete they are complementary. > > > > --- "Olle E. Johansson" <[EMAIL PROTECTED]> wrote: > > > > > My main question lies in the interworking between iptel's SER and > > Asteriks. > > > Not only on the configuration side, but also on the network side > > (here I > > > mean: can both run on the same server, or do they need to have > > different IP > > > addresses, ...). > > My 10 cents: > > > > Make sure that you run the two SIP proxies (SER and Asterisk) on > > different port numbers if you share one IP address. > > > > Why not make it simple and only run Asterisk? Is it a very large > > family :-) > > > > /O > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > = > Chris Albertson > Home: 310-376-1029 [EMAIL PROTECTED] > Cell: 310-990-7550 > Office: 310-336-5189 [EMAIL PROTECTED] > KG6OMK > > __ > Do you Yahoo!? > The New Yahoo! Shopping - with improved product search http://shopping.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] WARNING[49159]
I don't see any warnings in your trace. regards Martin On Tue, 14 Oct 2003, listas iPfone wrote: > Hi Martin! > > here is: > > s="Tue, 14 Oct 2003 17:55:00 GMT", ;expires=3600 > Expires: 159 > Content-Length: 0 > > > 9 headers, 0 lines > 11 headers, 0 lines > Reliably Transmitting: > REGISTER sip:sip.microcity.com.br SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983 > From: ;tag=as119e76aa > To: > Call-ID: [EMAIL PROTECTED] > CSeq: 199 REGISTER > User-Agent: Asterisk PBX > Expires: 160 > Contact: > Event: registration > Content-length: 0 > > (no NAT) to 200.251.160.60:5060 > Sip read: CLI> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983 > From: ;tag=as119e76aa > Call-ID: [EMAIL PROTECTED] > CSeq: 199 REGISTER > Server: Intertex ix66-release-2-0-4 > To: > Content-Length: 0 > > > 8 headers, 0 lines > Sip read: CLI> > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983 > From: ;tag=as119e76aa > To: ;tag=as0ebd4a9a > Call-ID: [EMAIL PROTECTED] > CSeq: 199 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Proxy-Authenticate: Digest realm="asterisk", nonce="5347caf4" > Content-Length: 0 > > > 11 headers, 0 lines > 12 headers, 0 lines > Reliably Transmitting: > REGISTER sip:sip.microcity.com.br SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983 > From: ;tag=as119e76aa > To: > Call-ID: [EMAIL PROTECTED] > CSeq: 200 REGISTER > User-Agent: Asterisk PBX > Proxy-Authorization: Digest username="miklos", realm="asterisk", > algorithm="MD5", uri="sip:[EMAIL PROTECTED]", nonce="5347caf4", > response="5af102c602abf311b8ec4c4eac72" > Expires: 160 > Contact: > Event: registration > Content-length: 0 > > (no NAT) to 200.251.160.60:5060 > Sip read: CLI> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983 > From: ;tag=as119e76aa > Call-ID: [EMAIL PROTECTED] > CSeq: 200 REGISTER > Server: Intertex ix66-release-2-0-4 > To: > Content-Length: 0 > > > 8 headers, 0 lines > Sip read: CLI> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983 > From: ;tag=as119e76aa > To: ;tag=as0ebd4a9a > Call-ID: [EMAIL PROTECTED] > CSeq: 200 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Expires: 160 > Contact: ;expires=160 > Date: Tue, 14 Oct 2003 17:24:44 GMT > Content-Length: 0 > > > 12 headers, 0 lines > Sip read: CLI> > REGISTER sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.33:5060 > Call-ID: [EMAIL PROTECTED] > Contact: "35" > CSeq: 26288 REGISTER > From: ;tag=0952-f0f0f9a2 > Supported: timer > To: "35" ;tag=as2f9c027e > Proxy-Authorization: Digest > username="35",realm="asterisk",uri="sip:[EMAIL PROTECTED]",nonce="15dd12ff",re > sponse="e94ab01fd7e8aff59a9b787f2f2a9288" > User-Agent: ipDialog SipTone 1.2.0 rc V UA > Expires: 3600 > Content-Length: 0 > > > 12 headers, 0 lines > Using latest request as basis request > Sending to 192.168.0.33 : 5060 (non-NAT) > Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.0.33:5060 > From: ;tag=0952-f0f0f9a2 > To: "35" ;tag=as2f9c027e > Call-ID: [EMAIL PROTECTED] > CSeq: 26288 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Content-Length: 0 > > > to 192.168.0.33:5060 > Transmitting (no NAT): > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.0.33:5060 > From: ;tag=0952-f0f0f9a2 > To: "35" ;tag=as2f9c027e > Call-ID: [EMAIL PROTECTED] > CSeq: 26288 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Proxy-Authenticate: Digest realm="asterisk", nonce="423ebf17" > Content-Length: 0 > > > to 192.168.0.33:5060 > Sip read: CLI> > REGISTER sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.33:5060 > Call-ID: [EMAIL PROTECTED] > Contact: "35" > CSeq: 26289 REGISTER > From: ;tag=483a-f0f0b8ca > Supported: timer > To: "35" > Proxy-Authorization: Digest > username="35",realm="asterisk",uri="sip:[EMAIL PROTECTED]",nonce="423ebf17",re > sponse="31561f3026dd65147adf46891c6b8129" > User-Agent: ipDialog SipTone 1.2.0 rc V UA > Expires: 3600 > Content-Length: 0 > localhost*CLI> > > 12 headers, 0 lines > Using latest request as basis request > Sending to 192.168.0.33 : 5060 (non-NAT) > Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.0.33:5060 > From: ;tag=483a-f0f0b8ca > To: "35" ;tag=as3028bf6d > Call-ID: [EMAIL PROTECTED] > CSeq: 26289 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Content-Length: 0 > > > to 192.168.0.33:5060 > Transmitting (no NAT): > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.0.33:5060 > From: ;tag=483a-f0f0b8ca > To: "35" ;tag=as3028bf6d > Call-ID: [EMAIL PROTECTED] > CSeq: 26289 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Ex
[Asterisk-Users] Mitel 5055 phone
Hello, I have seen the Mitel 5055 SIP phone mentioned a few times on the list, does anyone have any wonderful or horrible things to say about it? We are thinking about using them because they have many more programmable buttons than the Snom200 phones and are about $70 cheaper. Thanks, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] managers.conf Clarification Question
Does Anyone have a breakdown on what each option means in manager.conf system,call,log,verbose,command,agent,user I want to make a user who does not get a ton of events in the socket and is just for sending a query and getting that 1 reply I dont want to keep restarting my pbx to figure it out. I'm sure some may be self-explanatory but I was wondering if anyone knows for sure which options are which. Do you Yahoo!? The New Yahoo! Shopping - with improved product search
Re: [Asterisk-Users] */SER/FW
I think the way to run SER and * on the same box is to have Asterisk listen to SIP only on the loopback interface 127.0.0.1 that way no SIP clients will ever connect to Asterisk. Configure SER to use all the interfaces. SER will connect to SIP clients over the external interface and will notice Asterisk on 127.0.0.1 Configure SER to route SIP to SIP calls and forward the rest to Asterisk. The details depend on what you are trying to do and maybe even the caller IDs of the SIP clients? It just depends. In my case I'd like any SIP user in the world to be able to call me just like any PSTN user in the world can call me but I also would like to provide a service to freinds and family that would let them make SIP calls to each other, no need to involve Asterisk in that. I don't want the audio data going over my DSL link. The reason one might want to use SER is because it can do things with SIP that Asterisk can't. SER and Asterisk don't compete they are complementary. --- "Olle E. Johansson" <[EMAIL PROTECTED]> wrote: > > > My main question lies in the interworking between iptel's SER and > Asteriks. > > Not only on the configuration side, but also on the network side > (here I > > mean: can both run on the same server, or do they need to have > different IP > > addresses, ...). > My 10 cents: > > Make sure that you run the two SIP proxies (SER and Asterisk) on > different port > numbers if you share one IP address. > > Why not make it simple and only run Asterisk? Is it a very large > family :-) > > /O > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
