Re: [asterisk-users] call-limit in database
Dear I am using this function with L for example in the dbase. app=Dial appdata=SIP/[EMAIL PROTECTED]|60|L(10) it means dial 1 thru 1.1.1.1, with limitation=10 mili-second, and time out=60 sec best Mani --- Bhrugu Mehta [EMAIL PROTECTED] wrote: hi, all proble: I have add CALL-LIMIT field in my sip table in mysql. but when i call using sip same error occurred when use simple sip.conf file. is this possible to add CALL-LIMIT field in sip realtime table in mysql. if yes than how Bhrugu Mehta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bulk Reverse Phone Lookup
I've used http://www.555-1212.com, but not at the volume you're talking about. Maybe you can work a deal with them. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Norman Franke Sent: Wednesday, December 19, 2007 3:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bulk Reverse Phone Lookup Is anyone aware of a service where we can lookup phone numbers to determine a name and/or name + address available in bulk? We want to look up every number called to our call center, so it will be tens of thousands per day. Services that charge 3 to 5 cents per lookup will get way too expensive very quickly. Thus, I'm looking for a service that can either license a database or provide bulk lookups for maybe $300-$500/mo? Or even license a database for a few grand. Anyone know of something like this? -Norman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Summary: Upgrading to Asterisk 1.4
We've started testing Asterisk 1.4 1.2 has been very stable and we
have processed millions of minutes with it, SIP-to-ZAP, SIP-to-SIP and
SIP-to-IAX. We've been using Asterisk over 4 years now and it has
really re-invented the way me and a few others think of telephones.
The only inter-op issues I've ever seen are with JerkJerk's NuFone
network when they were insisting on using CVS-HEAD. How things have
changed. Right now one of our main gateways runs 1.0 something its
internal only passes calls from IAX to ZAP and it just works -- why
change it?
I have not dug into any 1.4 features yet except for T38 passthrough
and it seems to work ok. I need to submit a bugreport for some issues
next time I come across them.
{emphasis added}What are the plans for Asterisk 1.6 in regards to
furthering T.38 support?{/emphasis added}
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Re: [asterisk-users] Summary: Upgrading to Asterisk 1.4
22 dec 2007 kl. 10.55 skrev Andrew Joakimsen:
We've started testing Asterisk 1.4 1.2 has been very stable and we
have processed millions of minutes with it, SIP-to-ZAP, SIP-to-SIP and
SIP-to-IAX. We've been using Asterisk over 4 years now and it has
really re-invented the way me and a few others think of telephones.
The only inter-op issues I've ever seen are with JerkJerk's NuFone
network when they were insisting on using CVS-HEAD. How things have
changed. Right now one of our main gateways runs 1.0 something its
internal only passes calls from IAX to ZAP and it just works -- why
change it?
Right!
I have not dug into any 1.4 features yet except for T38 passthrough
and it seems to work ok. I need to submit a bugreport for some issues
next time I come across them.
yes, please.
{emphasis added}What are the plans for Asterisk 1.6 in regards to
furthering T.38 support?{/emphasis added}
There's a lot of work going on. The initial code base we got wasn't
well architectured in the signalling parts (chan_sip) and Josh Colp
is working on a rewrite. In the bug tracker, we have code additions
for asterisk-addons (due to licensing of spandsp libraries) to
create fax and t.38 send/receive applications, making Asterisk a
full T.38 endpoint and gateway. It is really exciting news, but will
take time to get there.
Merry Christmas!
/Olle
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Re: [asterisk-users] [asterisk-biz] Trixbox Phones Home
We expect Kerry Garrison to respond to this live Friday 21st Dec at 12 Noon EST with what steps they are taking and why. http://VoipUsersConference.org IRC: #voip-users-conference on Freenode.net Thanks to all who participated in the call. A lot of interesting side issues came up such as who should make money on what. It always amazes me the distance between the diametrically opposed viewpoints but I think we can all agree that we wish the entire asterisk community a great 2008 and a Wonderful, Frank Cappa-esque life The mp3 recordings of all calls are available in a list here: http://food4wine.ning.com/conference Happy, Prosperous and meaningful New Year to all! Next week, VOIP 2007 in review. January Conference Highlights: Jan 4th, Mark Spencer mid-January: Junction Networks randy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bulk Reverse Phone Lookup
Take a look at www.411xml.com On Dec 19, 2007 4:35 PM, Norman Franke [EMAIL PROTECTED] wrote: Is anyone aware of a service where we can lookup phone numbers to determine a name and/or name + address available in bulk? We want to look up every number called to our call center, so it will be tens of thousands per day. Services that charge 3 to 5 cents per lookup will get way too expensive very quickly. Thus, I'm looking for a service that can either license a database or provide bulk lookups for maybe $300-$500/mo? Or even license a database for a few grand. Anyone know of something like this? -Norman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Mon, Dec 17, 2007 at 10:40:32PM +0100, Benny Amorsen wrote: Olle E Johansson [EMAIL PROTECTED] writes: But on the other hand, if people rely on third-party distributions we might want to set up some kind of peer pressure on the maintainers - and possibly identify them so we can support them and speed up their process. Third-party distributions are very important, and Asterisk has for various reasons done relatively badly there. Fedora still doesn't have Asterisk, but does have CallWeaver. Asterisk isn't even available in the most popular extra repositories, but only in ATrpms, my least favourite of the larger repositories. It happens to be my favourite thrid party repo though, ;) and indeed there is quite some asterisk support happening there. -- Axel.Thimm at ATrpms.net pgpxYdtxsy9Yh.pgp Description: PGP signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
For the price Grandstream GXP-2000 is very feature packed and has a decent size and resolution display. The menus aren't the nicest but the phone works and it does not sound bad. For $70 you get what you pay for and the firmware is pretty stable and always being updated. On Dec 19, 2007 11:33 PM, d tbsky [EMAIL PROTECTED] wrote: Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
http://spc.pifiu.com for the stuff Linksys are Nazis about. On Dec 21, 2007 1:56 PM, Igor A. Goncharovsky [EMAIL PROTECTED] wrote: Hi! d tbsky wrote: ok. i will add linksys to our testing list. but cisco tend to lock things. can we get firmware for linksys easily ? or we must pay like cisco routers and switches? You can download latest firmware from linksys.com, also here is firmware release notes with full changes list. There is some support issues: support of VoIP devices only for itsp, but community can give answer on very-very advanced questions. -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sounds transscript / speech synthesis
Hi, in the earlier version there was a sounds.txt with the transcript of the soundfiles. Does this still exist somewhere? Is there a plan to make speech synthesis available the same way as soundfiles, ie. instead of playing language/soundfile.wav, send the text to the speechengine and play the output...? Jay... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] On-the-phone
Is there anyway to code in the Asterisk dialplan to show BOTH lines are busy when either of 200 or 201 are in use? exten = 200,hint,SIP/200SIP/201 exten = 201,hint,SIP/200SIP/201 Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
On Sat, 22 Dec 2007, Andrew Joakimsen wrote: For the price Grandstream GXP-2000 is very feature packed and has a decent size and resolution display. The menus aren't the nicest but the phone works and it does not sound bad. For $70 you get what you pay for and the firmware is pretty stable and always being updated. I've been a bit of a Grandstream enthusiast for some time now, and generally get on very well with them, but I have to say that recently I've become somewhat irritated by them. I know they had some hardware issues way back, but I've never encountered any of them. For the money, it *is* a feature packed phone. It's a bit plasticy and the sound quality isn't as good as say a Snom, but I've a lot of customers for whom the price is right. For a long time I've used software version 1.1.1.14, but recently they've had newer versions which I've been unable to downgrade to that version. Their latest version, 1.1.5.15 has fixed a few things, removed the router function (which I had no use for anyway), but in return I now get the occasional buzzing sound every few minutes from one older phone I've upgraded. Not sure about the new ones, not dared to upgrade them yet! Additionally, some phones I have out in the field which were shipped with some intermediate version of the s/ware have odd audio problems. I've emailled them, and get replies eventually, nothing forthcoming about a fix yet, so who knows... I do like the phones though, the GXP2000 is very nice, if a little fisher price, and for the money I can't better it, but if their software doesn't stabilise, then who knows what I'll be tempted to move to in the future... (probably Snom if I can persuade my customers to pay a little more - the 300's screen is jsut too small, and the price difference between that and the 320 is just a bit too much for my market...) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Summary: Upgrading to Asterisk 1.4
On Saturday 22 December 2007 01:51:56 am Johansson Olle E wrote: With that, I'm now changing my focus from SIP invite states, RTP sessions and video formats to Christmas ham purchasing, baking Christmas bread (julvört) and decorating the Christmas tree. Of course, you understand that there's an Asterisk asterisk on top of all those trees, right? :-) Merry Christmas! And thank you. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Send SIP 100 Trying instead of 183 Session Progress
You're right of course. I should have dug into this a little deeper and checked to see if it is corrected in the current release. As is so often the case, I was working on a real specific problem and once the system started doing what I wanted it to I pretty much forgot about it. I won't drag this out but I do want to clarify one point. This scenario occurs when the call originates from the SIP side and is destined for a number on the PRI side. The PRI does the trunk setup and then the SIP 183 is sent back to the originator before any further call progress occurs on the PRI side. This results in the SIP originator seeing a ring and then a busy if the called party is actually busy. Not deadly (although the original poster seemed to have some equipment that didn't like it) but certainly irritating. Richard On Dec 22, 2007, at 2:23 AM, Johansson Olle E wrote: 21 dec 2007 kl. 22.24 skrev Richard Revels: You are probably running into the problem described below. Below that is a link to the original document with the code patch. I put it on a PRI box we use inhouse and it took care of the 183 before a busy for me. However, this is a box we use inhouse. I've never put it on anything in production. Your mileage may vary gday guys (n'gals). I have a third party SIP platform which generates outbound calls via asterisk to ISDN (Australia - so thats ETSI ISDN). This platform doesn't really like inband signalling on outbound calls (ie getting 183's with SDP -- its fine with 180 Ringing etc...) Having had a bit of a silly time with the sip.conf variable progressinband=never,no,yes (arg!) I dug a little deeper into the chan_sip code. It appears on a SIP-Zap call the ISDN channel is opened, and before you can say 'boo' sip_write() in chan_sip is called this appears to occurs prior to any ISDN signalling (such as PRI_EVENT_PROCEEDING etc..) sip_write doesn't seem to care at all what progressinband is set to, and if it gets a frame when the SIP channel is not in AST_STATE_UP it generates a 183 with SDP (then sets SIP_PROGRESS_SENT) Does this behaviour seem strange? I'm not really sure if this is a bug, a 'its just like that' thing, or something strange with our ISDN that is unusual? In an ideal world (for me anyway... *grin*) I would think that progressinband=never (or even progressinband=no) would mean that 180 Ringing, 486 Busy etc would be used and 183 Session Progress with SDP would not... I don't think progressinband controls early media (audio to caller before call setup) but how indications should be sent (in audio=inband). If we get early media from the callee leg of the call, we have to relay it always. If you get early media signalling in SIP and don't have early media on the outbound call leg, then there's a bug and you should open a bug in the bug tracker so we can resolve it. For license reasons, we can't handle patches on the mailing list, we have to get them through the bug tracker. I really appreciate your help in resolving this issue, as you clearly have a lot of insight in the situation. Please open a bug on the bug tracker and we'll meet you there! Thanks, /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * The Asterisk SIP Masterclass - Stockholm, Sweden, January 2008 * Register today! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP handling - why 491 Request Pending response
Hi, The message that asterisk receives is not a retransmission but this is the same message but it enters asterisk from other sip proxy which is not a loop. The flow is the following Asterisk SIP Proxy (Location Service) INVITE (to registrar) - INVITE (to voicemail when not registered) when message enters asterisk for the second time it ofcorse has some extra SIP specific header like Record-Route and Via and the Request-URI is changed. And this causes 491 response. Can I do something about this? Can this behaviour be controlled, what do I have to change in the message so that asterisk won't treat it with 491 response? Thanks Tomasz On Dec 21, 2007 7:28 PM, Terry Wilson [EMAIL PROTECTED] wrote: What is the reason for such response? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0 ;received=192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74 ;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP 192.168.129.74:5070;branch=z9hG4bK5b33ae78;rport=5070 From: IPFon sip:[EMAIL PROTECTED]:5070;tag=as7217acbc To: sip:[EMAIL PROTECTED];tag=as7217acbc Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Asterisk will send a 491 Request Pending when it is currently processing an INIVTE on a particular call and it gets another INVITE that isn't a retransmission. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Summary: Upgrading to Asterisk 1.4
Hi!
Now over to a summary of the feedback. I'm not going deeper into bugs
reported, those will be handled separately.
Looks like I am a bit late, but I'll try to add my share as well to
highlight some of the issues that are invovled with 1.2 to 1.4
transition:
- with the advent of the g726aal2 troubles my preferred codec was
rendered unusable, and it still is that way because this setup is too
flakey, you never know if and when garbled audio will hit you. This still
does not work cleanly between 1.2 and 1.4 Asterisk boxes, with me
thinking that somehow on IAX this is more troublesome than on SIP. Only
alaw/ulaw (too hungry) and gsm (too sparse) are left since ilbc has the
potential to crash asterisk once a while (not always, not on every box).
- likewise SIP INFO DTMF worked reasonable well in Asterisk 1.2, whereas
my experience is that in 1.4 one should better move (back) over to
RFC2833, and when doing so don't forget about the rfc2833compensate
setting.
- all the transitions of the type application -- function can be
painful and error prone, especially for what concerns the replacements
for DBPut and DBGet and all the levels of () and [] and {} that are now
invovled.
- the GROUP_COUNT and call-limit (SIP) features saw a *lot* of changes on
their path from 1.0 to 1.2 to 1.4, and I hear that for 1.6 call-limit
will be touched and changed yet again. So practically every new point
release does this in an entirely different fashion.
By the way, the README file in asterisk-1.4 is outdated and refer to
upgrade instructions from 1.0 to 1.2.
Having said all of the above: Asterisk is coool and great, and
everyone involved even more so - Olle included ;-) - thank you for all
the effort!
Cheers happy days,
Philipp von Klitzing
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Re: [asterisk-users] Asterisk SIP handling - why 491 Request Pending response
22 dec 2007 kl. 15.51 skrev Tomasz Zieleniewski: Hi, The message that asterisk receives is not a retransmission but this is the same message but it enters asterisk from other sip proxy which is not a loop. The flow is the following Asterisk SIP Proxy (Location Service) INVITE (to registrar) - INVITE (to voicemail when not registered) when message enters asterisk for the second time it ofcorse has some extra SIP specific header like Record-Route and Via and the Request-URI is changed. And this causes 491 response. Can I do something about this? Can this behaviour be controlled, what do I have to change in the message so that asterisk won't treat it with 491 response? Without seeing the full SIP dialog, there's not much I can do or say. I would say that it would be better if the proxy could reply with an error message and that you used Asterisk to forward to voicemail when it gets that error message. /Olle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call-limit in database
call-limit is to set number of alternate calls . and L is to limit duration of each call . On Dec 22, 2007 2:54 PM, Pezhman Lali [EMAIL PROTECTED] wrote: Dear I am using this function with L for example in the dbase. app=Dial appdata=SIP/[EMAIL PROTECTED]|60|L(10) it means dial 1 thru 1.1.1.1, with limitation=10 mili-second, and time out=60 sec best Mani --- Bhrugu Mehta [EMAIL PROTECTED] wrote: hi, all proble: I have add CALL-LIMIT field in my sip table in mysql. but when i call using sip same error occurred when use simple sip.conf file. is this possible to add CALL-LIMIT field in sip realtime table in mysql. if yes than how Bhrugu Mehta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Summary: Upgrading to Asterisk 1.4
At 01:51 12/22/2007, Johansson Olle E wrote: Friends, We might have to reconsider our support policy here, where we developers abandoned 1.2 this summer. We might need another team that runs 1.2 support in the bug tracker. Pretty please, with cranberry sauce on top. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Summary: Upgrading to Asterisk 1.4
With that, I'm now changing my focus from SIP invite states, RTP sessions and video formats to Christmas ham purchasing, baking Christmas bread (julvört) and decorating the Christmas tree. Of course, you understand that there's an Asterisk asterisk on top of all those trees, right? :-) After Christmas, I'm running the new Asterisk SIP Masterclass together with Daniel Mierla here in Stockholm. He's one of the core OpenSER developers and it's going to be a great class. I'm sure we will locate a set of new interesting bugs in svn trunk during that week. I'm really looking forward to that training. (Hint: We still have a few open seats... :-) ) Greetings from a dark and cold place in Sweden, without a decent amount of snow... Have a wonderful, merry and cheerful Christmas! /Olle Merry Christmas to all on the list and thank you. Tony Plack ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Summary: Upgrading to Asterisk 1.4
Andrew Joakimsen wrote:
{emphasis added}What are the plans for Asterisk 1.6 in regards to
furthering T.38 support?{/emphasis added}
If you really want further T.38 support, then you should be looking at
callweaver. (An Asterisk 1.2 branch).
The T.38 support appears to be a lot better than the available
documentation suggests.
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Re: [asterisk-users] Asterisk SIP handling - why 491 Request Pending response
You can not do this. You can not have an INVITE that Asterisk originated enter back into Asterisk. Technically this is not a loop, but this is an INVITE glare and the way Asterisk is reacting is correct. You'll need to change the Call-Id of the INVITE that goes into Asterisk (a proxy can not do that so you'll need a B2BUA), or else you can do something like what Olle suggested. Thanks, Raj On Dec 22, 2007 9:51 AM, Tomasz Zieleniewski [EMAIL PROTECTED] wrote: Hi, The message that asterisk receives is not a retransmission but this is the same message but it enters asterisk from other sip proxy which is not a loop. The flow is the following Asterisk SIP Proxy (Location Service) INVITE (to registrar) - INVITE (to voicemail when not registered) when message enters asterisk for the second time it ofcorse has some extra SIP specific header like Record-Route and Via and the Request-URI is changed. And this causes 491 response. Can I do something about this? Can this behaviour be controlled, what do I have to change in the message so that asterisk won't treat it with 491 response? Thanks Tomasz On Dec 21, 2007 7:28 PM, Terry Wilson [EMAIL PROTECTED] wrote: What is the reason for such response? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0 ;received=192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74 ;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP 192.168.129.74:5070;branch=z9hG4bK5b33ae78;rport=5070 From: IPFon sip:[EMAIL PROTECTED]:5070 ;tag=as7217acbc To: sip:[EMAIL PROTECTED];tag=as7217acbc Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Asterisk will send a 491 Request Pending when it is currently processing an INIVTE on a particular call and it gets another INVITE that isn't a retransmission. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on IBM Netvista 2800 8364-EXX?
On Thu, 13 Dec 2007 20:40:08 -0600, Michael Graves [EMAIL PROTECTED] wrote: One of the major advantages of using voip is that call termination and DIDs are wholly separate matters. You can send outbound calls to various ITSPs based on least cost routing, leaving your POTS lines free to take incomming calls. The flexibility truly is worth the small extra cost. If I find a solid DSL connection + ITSP, and I need the independence of moving the server around, I might well port our number to an ITSP. That still leaves the issue of the lack of quality of voice connections over the Net. In the mean time, a regular POTS line is much more reliable. BTW, I finally got the IBM Netvista to boot :-) While the CF cards are on their way, I'd like to find an Asterisk distro that can run on a diskless station, so I can check whether the OpenVox TDM card works OK, and that voice quality is OK on such small hardware. I'll create a new thread on this. Still using your embedded Asterisk? http://www.smallnetbuilder.com/index2.php?option=com_contenttask=viewid=24210pop=1page=0Itemid=72 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PXE-bootable diskless Asterix distro?
Hello Since I got the IBM Netvista to boot Linux, and am still waiting for the Compact Flash cards that I ordered, I was wondering if someone knew of an Asterisk distribution that can run on that kind of diskless host? I've taken a look at AstLinux and AskoziaPBX, but they both seem to be meant to be installed on a solid-state medium instead of RAM. For instance, the Netvista is unable to uncompress a kernel, so expects linux instead of vmlinuz, etc. Or is it trivial even for a non-guru like me to modify such light Asterisk distros to run diskless? Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on IBM Netvista 2800 8364-EXX?
On Sun, 23 Dec 2007 02:29:13 +0100, Vincent wrote: On Thu, 13 Dec 2007 20:40:08 -0600, Michael Graves [EMAIL PROTECTED] wrote: One of the major advantages of using voip is that call termination and DIDs are wholly separate matters. You can send outbound calls to various ITSPs based on least cost routing, leaving your POTS lines free to take incomming calls. The flexibility truly is worth the small extra cost. If I find a solid DSL connection + ITSP, and I need the independence of moving the server around, I might well port our number to an ITSP. That still leaves the issue of the lack of quality of voice connections over the Net. In the mean time, a regular POTS line is much more reliable. This tends to be overblown. I found that the problems were entirely within my control and not inherent in sending calls over the internet. In fact, given the trouble I had with early FXO interfaces land lines were less reliable. BTW, I finally got the IBM Netvista to boot :-) While the CF cards are on their way, I'd like to find an Asterisk distro that can run on a diskless station, so I can check whether the OpenVox TDM card works OK, and that voice quality is OK on such small hardware. I'll create a new thread on this. Still using your embedded Asterisk? http://www.smallnetbuilder.com/index2.php?option=com_contenttask=viewid=24210pop=1page=0Itemid=72 Yes, still using Astlinux and loving it. Looking forward to the 0.50 release that should any day now. I'm also toying with Askozia (www.askozia.com) which combines Asterisk and the m0n0wall GUI on FreeBSD. Both are good choices for diskless installations. Astlinux is a little more flexible and mature, but lacking a serious GUI. Askozia has the GUI, but is still maturing and adding features. GUIs are like that, the underlying technology is a full Asterisk installation, but only so much of it is exposed for configuration via the GUI. BTW, I highly recommend the VOIP Users Conference at http://www.voipusersconference.org/ning/. Several of the conference members are using Asterisk embedded systems. It's an interesting group and the calls have been very instructional. Michael -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PXE-bootable diskless Asterix distro?
On Sun, 23 Dec 2007 02:34:45 +0100, Vincent wrote: Hello Since I got the IBM Netvista to boot Linux, and am still waiting for the Compact Flash cards that I ordered, I was wondering if someone knew of an Asterisk distribution that can run on that kind of diskless host? I've taken a look at AstLinux and AskoziaPBX, but they both seem to be meant to be installed on a solid-state medium instead of RAM. For instance, the Netvista is unable to uncompress a kernel, so expects linux instead of vmlinuz, etc. Or is it trivial even for a non-guru like me to modify such light Asterisk distros to run diskless? I'm not at all certain what you need to change on the hardware, but it seems to me it should be trivial. Perhaps something in the BIOS? I've used both H-P and Neoware hardware with no problem at all. I usually just set the BIOS to boot from a USB attached disk, then burn a USB key from and Astlinix or Askozia image. There's nothing all that special about the hardware. It's just a PC without a disk. PXE booting is really about having the boot image remoted from the host. That might be convenient for some people because can switch out or edit boot images easily. To my mind it has a a lot of administrative overhead. But I'm no Linux guru. I find it easier to swap out USB boot keys. Michael -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PXE-bootable diskless Asterix distro?
On Sun, Dec 23, 2007 at 02:34:45AM +0100, Vincent wrote: Hello Since I got the IBM Netvista to boot Linux, and am still waiting for the Compact Flash cards that I ordered, I was wondering if someone knew of an Asterisk distribution that can run on that kind of diskless host? Yes. I have a version of our CD that boots from PXE. It took minor changes and rebuilding as a PXE image, as Debian Live has basic support of that already. For simplicity I figure you'll be after a system that has everything in the initrd, but this is not the case here. It mounts a network partition to do the rest. We use NFS. CIFS is also supported. Debian Live takes a slightly different direction than astlinux and co.: Instead of completely rebuilding your system to match a read-only partition and read-write partition, just union-mount the read-write partition over the basic read-only partition. This technique is now common with live CDs. There are also ways to rewrite some changes, but I have not played with them at all, so I can't really say how effective they are. You can get the ISO from http://updates.xorcom.com/iso/ (the live ISO). Again, that is currently jsut a CD image, but it includes the full configuration for rebuilding. For more information about Debian Live: http://debian-live.alioth.debian.org/ . I'm currently using there casper as I have had problems getting initramfs-live on an Etch system (and had no time to solve them). But initramfs-live may be more useful to you. The really nice thing about that system is that it is a fully-functioning Debian system. Just apt-get install extra software (with the limitationsof your free RAM, of course). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'Traditional' Faxing
Depending on how many faxes you have coming in a simple fxs/fxo card will do the trick .. either Sagnoma or Digium or any others you could also use any decent ATA.. Asterisk only needs to know its a fax and what dialed number it came on to route it to the correct fax machine. Asterisk would just act as a pass thru.. Greg Cockburn wrote: Hi all, the company I work for has an aging Digital PBX attached to an E1. This PBX has a few analogue lines, one of which we use a 'traditional' fax machine on. I want to upgrade our PBX and Asterisk is almost a perfect fit. The only problem I can't seem to find a working solution for is Faxing. I don't want to use Hylafax or other similar methodologies. I believe there maybe someway to bridge an Analogue FXS port to a channel on the E1? Basically I want to mimic what we have now. 1. Any person can send a fax using the fax machine, and the PBX picks the next free channel on the E1. 2. A fax call can come over any channel on the E1, and the dialed number is matched and sent to the analogue FXS port of the PBX to be received by the fax machine. Is there anyway I can do this in Asterisk that will work seamlessly? I have not yet purchased any hardware, so recommendations would be greatly appreciated. (I believe some of the problem exists due to timing, does any hardware; E1 card / Analogue card; support linking a timing signal together?) Sangoma, Digium, Pika? Thanks all for any help on this one. Greg. --- *Text inserted by Panda IS 2008:* This message has NOT been classified as spam. If it is unsolicited mail (spam), click on the following link to reclassify it: It is spam! http://localhost:6083/Panda?ID=pav_11247SPAM=truepath=C:%5CDocuments%20and%20Settings%5Cshawn%5CLocal%20Settings%5CApplication%20Data%5CPanda%20Software%5CAntiSpam --- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP handling - why 491 Request Pending response
23 dec 2007 kl. 01.45 skrev Raj Jain: You can not do this. You can not have an INVITE that Asterisk originated enter back into Asterisk. Technically this is not a loop, but this is an INVITE glare and the way Asterisk is reacting is correct. You'll need to change the Call-Id of the INVITE that goes into Asterisk (a proxy can not do that so you'll need a B2BUA), or else you can do something like what Olle suggested. I don't really agree here Raj. Of course you can send an INVITE to an URI hosted by the proxy and the location table points back to one or several URI's in the same Asterisk server. /O ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
