RE: [asterisk-users] Unable to find a codec translation path from ilbcto ulaw
Oliver, SIP-phone --iLBC-- Asterisk ---ulaw PSTN-Gateway I get the following error: Unable to find a codec translation path from ilbc to ulaw Does your phone support ilbc as a codec ? Is the codec_ilbc loaded on the * box ? Usually you get this kind of error when the codec is not supported Stelios S. Koroneos Digital OPSiS - Embedded Inteligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Oliver Brandt Sent: Friday, April 27, 2007 7:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unable to find a codec translation path from ilbcto ulaw Hi! As the upstream of my DSL-connection is very slow, I'd like my sip-phones to use iLBC to connect to my *. My gateway provider only allows ulaw. Hence, I'd like to use the follwing setup: SIP-phone --iLBC-- Asterisk ---ulaw PSTN-Gateway I get the following error: Unable to find a codec translation path from ilbc to ulaw Setup SIP-phone: disallow=all allow=ilbc Setup PSTN-Gateway: disallow=all allow=ulaw I've googled for overn an houre. But no luck. So I'd really apreciate any help! Thanks! Oliver ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR changes in 1.4.3?
On 4/27/07, Steve Murphy [EMAIL PROTECTED] wrote: I'm the guilty party. I've been trying to fix several CDR bugs, involving stuff like missing times, missing changes in state (like NO_ANSWER when the call was ANSWERED), etc. A-HA! Don't get me wrong, I am not opposed to progress as there have been a few CDR quirks that were annoyances to me as well. The result is that several more cases are more accurate, but also, that rather uninteresting CDR's can be generated. In contemplating what could be done to get rid of some of these, I sometimes have to ask, is this truly something we have to get rid of?... In the meantime, uninteresting CDR's with NO_ANSWER and billsec=0, should be easy to filter out, right? You're right, they can/will be easy to filter out, I'll update my script that pulls CDR data for me to do it. I will, in the coming days, look at some of the extraneous CDR's that are generated, and see what I can do to get rid of them. It's not always that simple. If we ring a phone, for instance, and no-one answers it, is that truly, really, something that no-one will ever be, could ever be, interested in? (just a fer-instance). I do think there is a potential desire for some people to have these records. In my experience, however, unanswered calls are logged as well even before 1.4.3. Perhaps it is related to my trunk configuration. What is wholly uninteresting is, as you mentioned above, billsec/duration = 0 calls terminating to s in each context associated with the call and I'm not really sure what they could be used for but I'm sure somebody could find something. Perhaps a flag could be set to request regular, verbose, or very verbose CDR? Regular could provide behavior similar to pre-1.4.3, and verbose/very verbose could add more and more detail. Would it be simple enough to identify which CDRs are trivial and only log them when the verbosity is set higher? I know, that's easy for me to say, I'm not the one who has to code it up. :) I welcome your input. Complain up a storm. I'll try my best to make you happy. Thanks for the positive attitude. We do really appreciate the work you guys are doing, even if it doesn't seem like it at times. :) As I mentioned above, my only suggestion would be to identify CDRs that are informational in nature and only log them when a flag is set. Thanks again. sl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold issue with asterisk 1.4.2
Steve Finkelstein wrote on 4/28/07 12:21 AM: my musiconhold.conf: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 and finally in my extensions.conf: asterisk-1.4.2 # grep 100 /etc/asterisk/extensions.conf exten = 100,1,MusicOnHold(30) exten = 100,2,Hangup When I dial 100 however, I receive the following: [Apr 28 00:17:33] WARNING[30975]: res_musiconhold.c:947 local_ast_moh_start: No class: 30 The parameter to MusicOnHold is the class of music to play. You have no class named 30 just like the error says. :) You do have a class named default in the config snippet you pasted, so MusicOnHold(default) should work. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_CHANCONFIG failed onchannel1:Nosuchdeviceoraddress
On Sat, Apr 28, 2007 at 01:47:15PM +1200, CSB wrote: On Fri, Apr 27, 2007 at 07:11:48AM +1200, CSB wrote: [snip] As suggested earlier I replaced this with: /etc/modprobe.d/zaptel options wctdm opermode=NEWZEALAND honormode=1 boostringer=1 fastringer=1 [snip] dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.17.1 Zaptel Echo Canceller: KB1 wctdm: Unknown parameter `honormode' This is the problem Updated vi /etc/modprobe.d/zaptel options wctdm opermode=NEWZEALAND fxshonormode=1 boostringer=1 fastringer=1 Again, please: rmmod wctdm; modprobe wctdm ; dmesg | tail -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Two Connected Servers Sound Quailty
Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the best but still isn't reliable) The problems are intermittent sometimes the sound will cut out for 3-4 seconds and other times the sound will just be loosing every other word, and other times it sounds just fine. Also, we have been using Skype over this same Internet connection and have very good sound quality with very few lost words. So here are my questions. First, is it a correct assumption to say that because Skype works well over this connection then I should be able to get asterisk to work over this connect. I am hoping that Skype isn't better then asterisk in this area. If I should be able to get the same sound quality could you point me in the right direction on how to achieve this. (I have tried messing with the jitterbuffer but haven't been able to find very good docs on how to utilize this functionality so about all I have done is set jitterbuffer=yes) Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two Connected Servers Sound Quailty
Hi Matt, you didn't mention what type/bw of each site Internet connection, i suggest that you try to split the scenario into smaller pieces: - run long term pings between the server while you make a call and check for packet loss. - make internal calls between extensions on the same branch and verify that both servers work okay (eliminating Internet connectivity) - register a UA from one site to a server on the other site, make a call and viceversa (eliminating a problem on one of the servers). - check for speed/duplex setting on NIC and switch port. - check if the sound quality issues are symmetric (does both sides experience the sound cut or it only happens on a specific site). - make sure you don't use G.711 as it consumes bw and from the codec list you've mentioned has the lowest tolerance to packet loss. Since the problems are intermittent my bet is that someone in the office is have the p2p client work overtime or sending lots emails with funny attachments On 4/28/07, Matt Gardner [EMAIL PROTECTED] wrote: Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the best but still isn't reliable) The problems are intermittent sometimes the sound will cut out for 3-4 seconds and other times the sound will just be loosing every other word, and other times it sounds just fine. Also, we have been using Skype over this same Internet connection and have very good sound quality with very few lost words. So here are my questions. First, is it a correct assumption to say that because Skype works well over this connection then I should be able to get asterisk to work over this connect. I am hoping that Skype isn't better then asterisk in this area. If I should be able to get the same sound quality could you point me in the right direction on how to achieve this. (I have tried messing with the jitterbuffer but haven't been able to find very good docs on how to utilize this functionality so about all I have done is set jitterbuffer=yes) Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Free seating Agents and logged in / loggedoutindication
Yep it's possible though why not just use a handset with a microbrowser that states on the display logged in out or? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alexander Topolanek Sent: Saturday, 28 April 2007 1:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Free seating Agents and logged in / loggedoutindication Am Freitag, den 27.04.2007, 16:25 -0400 schrieb Dean Collins: A pop up on their pc display using Adhearsion to drive the resulting logged in/out popup? Sorry, I forgot to tell that the Agents don't have PC's. Is it possible to trigger the MWI from an AGI-script that is fired when an Agent is logged out? I would like to set up a call center with free seating agents. However I would like to indicate the agent status somehow on the terminal, to tell the agent if she has been logged out due to non-answer. -- Alexander Topolanek http://www.topolanek.at ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Two Connected Servers Sound Quailty
Try SIP if at all possible. I have had mixed results with IAX that SIP made go away. If you try SIP, you can at least rule out IAX as the cause. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yossi Ben Hagai Sent: Saturday, April 28, 2007 4:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Two Connected Servers Sound Quailty Hi Matt, you didn't mention what type/bw of each site Internet connection, i suggest that you try to split the scenario into smaller pieces: - run long term pings between the server while you make a call and check for packet loss. - make internal calls between extensions on the same branch and verify that both servers work okay (eliminating Internet connectivity) - register a UA from one site to a server on the other site, make a call and viceversa (eliminating a problem on one of the servers). - check for speed/duplex setting on NIC and switch port. - check if the sound quality issues are symmetric (does both sides experience the sound cut or it only happens on a specific site). - make sure you don't use G.711 as it consumes bw and from the codec list you've mentioned has the lowest tolerance to packet loss. Since the problems are intermittent my bet is that someone in the office is have the p2p client work overtime or sending lots emails with funny attachments On 4/28/07, Matt Gardner [EMAIL PROTECTED] wrote: Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the best but still isn't reliable) The problems are intermittent sometimes the sound will cut out for 3-4 seconds and other times the sound will just be loosing every other word, and other times it sounds just fine. Also, we have been using Skype over this same Internet connection and have very good sound quality with very few lost words. So here are my questions. First, is it a correct assumption to say that because Skype works well over this connection then I should be able to get asterisk to work over this connect. I am hoping that Skype isn't better then asterisk in this area. If I should be able to get the same sound quality could you point me in the right direction on how to achieve this. (I have tried messing with the jitterbuffer but haven't been able to find very good docs on how to utilize this functionality so about all I have done is set jitterbuffer=yes) Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New VICIDIAL astGUIclient Release: 2.0.3
Do you guys have an ISO install CD yet? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Friday, April 27, 2007 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] New VICIDIAL astGUIclient Release: 2.0.3 Hello, We've released another update to our astGUIclient suite: 2.0.3 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL call center suite. This package is free and GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have focused on fixing bugs and adding several new features including many new administrative functions and more campaign options. We have also tested the suite on Asterisk versions through 1.2.18. All client web-apps and administration pages are available in English, Spanish, Greek and German, with rough translations of French, Polish, Italian, Portuguese and Brazillian Portuguese for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New VICIDIAL astGUIclient Release: 2.0.3
Not yet, but it is something we are working on. There are a few people that have made some special-hardware ISOs for VICIDIAL but they are by no means universal, more for quick install on specific high-end servers. Right now we are just concentrating on making VICIDIAL as solid and feature-filled as possible. Thanks, MATT--- On 4/28/07, Steve Totaro [EMAIL PROTECTED] wrote: Do you guys have an ISO install CD yet? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Friday, April 27, 2007 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] New VICIDIAL astGUIclient Release: 2.0.3 Hello, We've released another update to our astGUIclient suite: 2.0.3 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL call center suite. This package is free and GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have focused on fixing bugs and adding several new features including many new administrative functions and more campaign options. We have also tested the suite on Asterisk versions through 1.2.18. All client web-apps and administration pages are available in English, Spanish, Greek and German, with rough translations of French, Polish, Italian, Portuguese and Brazillian Portuguese for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to find a codec translation path from ilbc to ulaw
Hi Dave! Thank you very much for replying! what gateway provider are you referring to?doesn't your sip phone webcalldirect (it does not seam to support iLBC directly) connect directly to * as your diagram indicated? Yes, my sipphone ist connected directly to * and also the gateway provider is directly connected to *. My * is on a root server at hosting provider (high bandwith internet connection to the gateway provider) but my phone is connected through DSL with a very limited upstream. For this reason I'd like asterisk to do the codec conversion from iLBC to ulaw. I bett all I have to do is load the codec or/and the codec translator for iLBC to ulaw. But when googleing I only find articles the describe, that * is doing the codec translation automatically. I can't find any information on how to load a codec or the translator manually. I'm probably just using the wrong search string in google... When * starts translators are beeing loaded, but as far as I can see non for iLBC to ulaw. I've put together another test setup with to sip phones to clarify the problem: [phone1] disallow=all allow=iLBC [phone2] disallow=all allow=ulaw When calling from one phone to the other I get the following message: chan_sip.c:2841 sip_call: No audio format found to offer. Cancelling call to phone2 Thank you very much again! Oliver ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fixed quantity calls per extension
Sorry if I´m not clear.
I´m using zap channels. I need to limit the number of calls that dial one
extension. No more than 3 calls using an IVR service (eagi) at the same
time.
May be It can be resolve using GROUP() and GROUP_COUNT()
exten = 99,1,Set(GROUP(99) = G99)
exten = 99,2,GotoIf($[${GROUP_COUNT(99)}3]?103)
exten = 99,3,eagi(Service1)
exten = 99,103,Hangup
On 4/27/07, Steve Edwards [EMAIL PROTECTED] wrote:
On Fri, 27 Apr 2007, Eric ManxPower Wieling wrote:
equis software wrote:
Hi, is there any way to configure a number of simultaneus calls per
extension.
I need to rerstrict the simultaneus calls per service ( in extension 33
I
answer Service 1 and in extension 37 I answer service 2.
Example:
No more than 3 simultaneus calls to extension 33
No more than 15 simultaneus calls to extension 37
Yes. Any of the following:
1) Check the documentation for your IP phone.
2) Use the applications shown by show applications like group in the
CLI.
3) Talk to your VoIP provider
4) Use Queues
Or, the question says simultaneus (sp) which could be interpreted as a
conference in which case meetme and meetmecount would do the trick.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Unable to find a codec translation path from ilbcto ulaw
Hi Stelios! Thank you very much for you reply! Does your phone support ilbc as a codec ? Definately. By using to phones and forcing them to use iLBC I can make calls from one phone to the other. The gateway provider does not support iLBC and so * has to do the conversion to ulaw. I've also put a test setup togther with to phones connected to *. One is beeing forced to use iLBC the other to use ulaw. I get the following message: [Apr 28 16:12:40] WARNING[25512]: chan_sip.c:2841 sip_call: No audio format found to offer. Cancelling call to fritz_kiel_oliver2 Is the codec_ilbc loaded on the * box ? I'm sure the problem is that the ilbc or/and the ulaw codec are not beeing loaded. I've googled for ever to find out how to load a codec but all I found were aticles telling me the * is doing the conversion automaticall. Do I also need to load a translator from ilbc to ulaw? How do I do that? Thank you very much again! Oliver ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Unable to find a codec translation path from ilbcto ulaw
Hi! As the upstream of my DSL-connection is very slow, I'd like my sip-phones to use iLBC to connect to my *. My gateway provider only allows ulaw. Hence, I'd like to use the follwing setup: SIP-phone --iLBC-- Asterisk ---ulaw PSTN-Gateway I get the following error: Unable to find a codec translation path from ilbc to ulaw First, do a 'show translation' in the asterisk console to confirm that asterisk is definitely not supporting the translation. If it doesn't support it, you'll see a '-' instead of a number in the column and row for ilbc. Debian's, and possibly other distributions binaries, seems to only support ilbc in passthrough mode because of 'non-free' issues with the codec. I assume there is a patent or something attached to it. I believe you can build your own Asterisk to get around the issue, and in fact I asked the same question a while back and I seem to remember (but I'm not sure) that someone pointed me to a pre-compiled version in someone else's Debian repository that included the ilbc transcoding stuff. hth James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Unable to find a codec translation path from ilbcto ulaw
Just to follow up on my previous comment, /usr/share/doc/asterisk/copyright contains the following on my Debian system: * The iLBC codec library code has been removed from the Debian asterisk package as it does not conform with the DFSG. James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Best Wireless bridge for Polycoms
If you are going to have clusters of phones like a cubicle setup, you could buy one of the Linksys routers like the WRT54G and setup WDS. Then plug four phones into it. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Friday, April 27, 2007 11:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Best Wireless bridge for Polycoms You can purchase the Linksys part PA100-NA and plug it into a WBP54G and then ignore the power connector hanging off the WBP54G. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: April 27, 2007 3:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Best Wireless bridge for Polycoms Michael and all those who replied, This Linksys WBP54G does seems to be what I need, but it also seems very much made for Linksys phones. Isn't there some sort of equivalent thing that comes with it's own power supply (at the cost of needing another outlet for the phone)? Alternatively, where do I find an adapter for NA power that turns into 2V 5A DC current? Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Friday, April 27, 2007 13:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best Wireless bridge for Polycoms --Original Message Text--- From: Mike Date: Fri, 27 Apr 2007 10:24:05 -0400 Hi, I'm stuck doing an install with Polycoms at a small office with no RJ-45. They went wireless 100%, poor them. I insist on using Polycom unless it's impossible because that's what I am standardized on for many reasons. What's the best way/device to turn a wired Polycom 501 (or any Polycom for that matter) into a WiFi phone? Mike Linksys makes a device spcifically for this role so that their SPA series IP phones can be connected to WIFI. http://www.linksys.com/servlet/Satellite?c=L_Product_C2childp agename=US%2FLayoutcid=1139961537989pagename=Linksys%2FCommo n%2FVisitorWrapperlid=3798954250B11 This device take power from the phone via a simple coax plug on the phone. Could easily be powered externally with a wall wart. Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] headsets for linksys/sipura phones?
Nabeel Jafferali wrote: You can look for headsets made for Motorola cell phones. Also, Plantronics has some compatible models - I can dig up part numbers if you're interested. Yes, please - Plantronics is in my regular suppliers catalog, but still only with 3.5mm jacks. If you've got part#s or URLs, that would be very helpful. /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] headsets for linksys/sipura phones?
Per Jessen wrote: Nabeel Jafferali wrote: You can look for headsets made for Motorola cell phones. Also, Plantronics has some compatible models - I can dig up part numbers if you're interested. Yes, please - Plantronics is in my regular suppliers catalog, but still only with 3.5mm jacks. If you've got part#s or URLs, that would be very helpful. Thanks for the excellent suggestion - I took a closer look at plantronics, and found the models with 2.5mm jacks. /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] headsets for linksys/sipura phones?
On 4/27/07, Per Jessen [EMAIL PROTECTED] wrote: Try your local mobile phone supplier. I used a headset that came with one of my cell phones, and it worked great w/ my SPA-941. Not a bad idea - which make was this for? None of my phones (Ericsson, Nokia) have a 2.5mm socket, they're all special/proprietary. The headset for any other mobile will work. And I thought Nokia did use 2.5mm but reverse polarity ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ADSL routers with integrated SIP QoS for other devices
Greetings list, Thanks to all who replied to my thread a few days ago SIP devices with packet loss tolerance. One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. I've used m0n0wall's QoS in the past with reasonable success, but it's quite a bulky and complex setup for deploying to remote sites which I'll never visit (minimum 3 boxes - ADSL modem, m0n0, WiFi AP). So, does anyone have any recommendations for a wireless ADSL router with integrated QoS for SIP/RTP? I've looked at some of the Draytek units (e.g. Vigor 2700V), but I can't find reference as to whether the integrated QoS applies only to the FXS ports in the router itself, or to all SIP traffic (most of the users will have separate SIP hardphones). These are all to be used in the UK, so the device in question needs to support PPPoA. Any suggestions gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two Connected Servers Sound Quailty
Hi Matt - I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the best but still isn't reliable) The problems are intermittent sometimes the sound will cut out for 3-4 seconds and other times the sound will just be loosing every other word, and other times it sounds just fine. First, is it a correct assumption to say that because Skype works well over this connection then I should be able to get asterisk to work over this connect. I am hoping that Skype isn't better then asterisk in this area. Yes, it's certainly possible to get good quality with asterisk. Skype is not better, they just build in more default latency. I've never measured exactly, but it seems that Skype calls typically have a built in buffer between 250ms and 1000ms. Asterisk, will try to use as little latency as possible. You've set jitterbuffer=yes, but you'll also need to set maxjitterbuffer (probably to 1000), resyncthreshold (probably to 1000), and maxjitterinterps (10 is a good safe value). You can try adjusting these values to see how it affects your calls. You'll also want to do something about QoS. If you don't, the next time you try to FTP a file, it will try to grab all your available bandwidth, whether or not you are on a call. This will surely screw up your call quality. If your routers have QoS options, you can ensure that your voice traffic will get first dibs on bandwidth. You'll need to configure QoS on both ends. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.14 will not run without internet connection
Hi Joseph - Thanks, I think you are on the right track. When no Sip adapters were connected to asterisk it took me over one minute from the time I typed reload to the time I've seen anything on the screen. When, I connected the all the sip devices and eliminated some entries in sip.conf and iax.conf it took only 22-seconds to reload Something else must be wrong. A reload should happen nearly instantaneously (less than 1 second), whether or not it can find your sip devices. You mentioned you're using asterisk 1.2.14. Can you describe the rest of your setup (OS, Hardware)? What asterisk modules and/or addon items are you using? - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold issue with asterisk 1.4.2
Interesting, that works David. I got the example directly out of the published VoIP Hacks book and followed instructions step by step. Either way, thanks much. :-) - sf Dave Miller wrote: Steve Finkelstein wrote on 4/28/07 12:21 AM: my musiconhold.conf: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 and finally in my extensions.conf: asterisk-1.4.2 # grep 100 /etc/asterisk/extensions.conf exten = 100,1,MusicOnHold(30) exten = 100,2,Hangup When I dial 100 however, I receive the following: [Apr 28 00:17:33] WARNING[30975]: res_musiconhold.c:947 local_ast_moh_start: No class: 30 The parameter to MusicOnHold is the class of music to play. You have no class named 30 just like the error says. :) You do have a class named default in the config snippet you pasted, so MusicOnHold(default) should work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail on Different Server
Hi Forest - I have two seperate systems at two different locations. Each hosts there own voicemail for their phones. I have thought about just having all voicemail on one server. Is the best way to do this just through a dial app? Can anyone think of draw backs to this? One I can think of is I will have to specify a extension to redirect 0 (for receptionist) back to the Site A server. I will also have to redirect all directory apps to the voicemail server. The first time I set up a multi-site Asterisk, I tried to do centralized voicemail. My only real motivation for it was to have a centralized directory. I originally did the NFS-mount method. If the internet connection ever went down at the non-central offices there were two problems 1) users didn't have access to their voicemail, 2) asterisk did not handle the situation gracefully. I believe asterisk will now handle the situation gracefully, but users still won't be able to get their voicemail. This was not acceptable to me or to the client, so I changed to storing the voicemail via ODBC on MySQL. Each server had it's own local storage, and then MySQL replicated the databases between the sites. This setup was terribly finicky and unstable. It was much worse than the NFS mount. I quickly gave it up. My eventual solution for this client was to store voicemail locally at each site. For the centralized directory, I just wrote a quick shell script to rsync the voicemail.conf file and all personal greetings between all the servers. Cron runs this script periodically to keep all the asterisk servers up to date. This solution is MUCH better. It's been very stable and reliable. There's now another option - you can store the messages on a central IMAP server. That may work for you. I haven't done this setup yet, but I believe if you wanted the centralized directory, you'd still need to do something like an rsync script between various asterisk servers. - Noah On 4/27/07, Anthony Rodgers [EMAIL PROTECTED] wrote: mount -o intr,nolock ought to do the trick. we're using those options now, but thankfully haven't had reason to find out if they work or not yet. CP Doug Garstang wrote: No, you can get Asterisk and NFS to work fine together. It was in my past job, so I can't remember the exact settings, but there was some magic combination of NFS client mount settings that would cause Asterisk to return immediately, rather than hang, if there was an NFS communications problem. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.14 will not run without internet connection
Noah Miller wrote: Hi Joseph - Thanks, I think you are on the right track. When no Sip adapters were connected to asterisk it took me over one minute from the time I typed reload to the time I've seen anything on the screen. When, I connected the all the sip devices and eliminated some entries in sip.conf and iax.conf it took only 22-seconds to reload Something else must be wrong. A reload should happen nearly instantaneously (less than 1 second), whether or not it can find your sip devices. You mentioned you're using asterisk 1.2.14. Can you describe the rest of your setup (OS, Hardware)? What asterisk modules and/or addon items are you using? - Noah Maybe some of the following observations might help. I have seen this when DNS is not working and there are register statements that use hostnames or FQDNs. That might have been a bug that was fixed though. I have also seen this on a server under very heavy load with IVR and queues and reload issued many times. A stop now and asterisk would bring it back to full speed. Again, this was a while ago, maybe that bug was fixed too. The third time I have seen it is if you are using the manager interface extensively, same as before, saw that maybe a year ago, might be fixed now. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail on Different Server
Noah Miller wrote: Hi Forest - I have two seperate systems at two different locations. Each hosts there own voicemail for their phones. I have thought about just having all voicemail on one server. Is the best way to do this just through a dial app? Can anyone think of draw backs to this? One I can think of is I will have to specify a extension to redirect 0 (for receptionist) back to the Site A server. I will also have to redirect all directory apps to the voicemail server. The first time I set up a multi-site Asterisk, I tried to do centralized voicemail. My only real motivation for it was to have a centralized directory. I originally did the NFS-mount method. If the internet connection ever went down at the non-central offices there were two problems 1) users didn't have access to their voicemail, 2) asterisk did not handle the situation gracefully. I believe asterisk will now handle the situation gracefully, but users still won't be able to get their voicemail. This was not acceptable to me or to the client, so I changed to storing the voicemail via ODBC on MySQL. Each server had it's own local storage, and then MySQL replicated the databases between the sites. This setup was terribly finicky and unstable. It was much worse than the NFS mount. I quickly gave it up. My eventual solution for this client was to store voicemail locally at each site. For the centralized directory, I just wrote a quick shell script to rsync the voicemail.conf file and all personal greetings between all the servers. Cron runs this script periodically to keep all the asterisk servers up to date. This solution is MUCH better. It's been very stable and reliable. There's now another option - you can store the messages on a central IMAP server. That may work for you. I haven't done this setup yet, but I believe if you wanted the centralized directory, you'd still need to do something like an rsync script between various asterisk servers. - Noah On 4/27/07, Anthony Rodgers [EMAIL PROTECTED] wrote: mount -o intr,nolock ought to do the trick. we're using those options now, but thankfully haven't had reason to find out if they work or not yet. CP Doug Garstang wrote: No, you can get Asterisk and NFS to work fine together. It was in my past job, so I can't remember the exact settings, but there was some magic combination of NFS client mount settings that would cause Asterisk to return immediately, rather than hang, if there was an NFS communications problem. Doug. Can you elaborate on this, I changed to storing the voicemail via ODBC on MySQL. Each server had it's own local storage, and then MySQL replicated the databases between the sites. This setup was terribly finicky and unstable. It was much worse than the NFS mount. I quickly gave it up. This sounds like it would probably work the best, especially if you have users moving around between offices. What was so finicky and unstable about it? I am not one to quickly give up. I have found that persistence pays off when the idea is sound. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ADSL routers with integrated SIP QoS for other devices
On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote: Thanks to all who replied to my thread a few days ago SIP devices with packet loss tolerance. One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. Sangoma S518 (internal PCI) on a Linux box with iproute2/iptables/tc or BSD with pf. These are the best solutions, IMO. The latest Linux kernels also have SIP connection tracking/matching, so it should be possible to mark packets and prioritize based on iptables matching. I have not done this just yet, as the latest 2.6.20/2.6.21 kernels do not play nice with the wanrouter drivers. (note: there was a recent patch to 2.6.20.4 which apparently has much better SIP matching, and has been tested successfully with Asterisk. I have not tested it yet, and the iptables guys have rejected the patch as their direction for packet matching is shifting significantly in the near future. It can be found at http://thread.gmane.org/gmane.comp.security.firewalls.netfilter.devel/18860.) I'm still looking for a miniPCI ADSL chipset that Linux can use, or an actual raw ADSL non-PCI chipset that I can design into an embedded system. If anyone has any leads, please don't hesitate to contact me! If you're curious, I have my rc.tc script for Linux up on http://mixdown.ca/~andrew/rc.tc. It's loosely based off of wondershaper, but works much better, IMO. It does host-based prioritization for VOIP, puts mail just underneath bulk traffic, and P2P beyond that (if you have the p2p connmark stuff set). I can completely saturate DSL links with the S518 with this config without appreciable VOIP degradation. Even without an S518, this script works well with external ADSL/cable modems. You may have to play with the upload rate; some cheap ADSL modems will start blocking your upstream traffic beyond as little as 50% of the upstream rate. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trixbox/FreePBX
Hello, Installed Trixbox with a digium card and it is taking 2 rings for it to pick up. Any suggestions how to have the system pickup immediately? Thanks, Neal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Trixbox/FreePBX
Hi, You need to post this on the trixbox forums.but as a fellow trixbox user I'll give you the answer. Turn off fax detection. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 28 April 2007 1:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Trixbox/FreePBX Hello, Installed Trixbox with a digium card and it is taking 2 rings for it to pick up. Any suggestions how to have the system pickup immediately? Thanks, Neal image001.gif Description: image001.gif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox/FreePBX
Hi, Write down your problem clearly. Thanks [EMAIL PROTECTED] wrote: Hello, Installed Trixbox with a digium card and it is taking 2 rings for it to pick up. Any suggestions how to have the system pickup immediately? Thanks, Neal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ADSL routers with integrated SIP QoS for otherdevices
Andrew wrote: On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote: Thanks to all who replied to my thread a few days ago SIP devices with packet loss tolerance. One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. Sangoma S518 (internal PCI) on a Linux box with iproute2/iptables/tc or BSD with pf. These are the best solutions, IMO. I was just about to reply with the same recommendation. A SFF chassis with 2 PCI slots could host one S518 and a PSTN interface. These units typically have built-in ethernet and some have built-in wireless. I still have my fingers crossed that Sangoma will offer an ADSL daughercard for the A200. That would make for a perfect combination in a SFF chassis... The latest Linux kernels also have SIP connection tracking/matching, so it should be possible to mark packets and prioritize based on iptables matching. I have not done this just yet, as the latest 2.6.20/2.6.21 kernels do not play nice with the wanrouter drivers. (note: there was a recent patch to 2.6.20.4 which apparently has much better SIP matching, and has been tested successfully with Asterisk. I have not tested it yet, and the iptables guys have rejected the patch as their direction for packet matching is shifting significantly in the near future. It can be found at http://thread.gmane.org/gmane.comp.security.firewalls.netfilter.devel/18 860.) I'm still looking for a miniPCI ADSL chipset that Linux can use, or an actual raw ADSL non-PCI chipset that I can design into an embedded system. If anyone has any leads, please don't hesitate to contact me! Good luck and let us know if you find one. The manufacturers of the XDSL chipsets seem to be even worse than the video card companies when it comes to OSS. There's a project on SF called OpenADSL that was working to make common XDSL chipsets work under Linux. The project appears almost dead with a developer post every 6~8 weeks, but that might be a good place to start Looking. If you're curious, I have my rc.tc script for Linux up on http://mixdown.ca/~andrew/rc.tc. It's loosely based off of wondershaper, but works much better, IMO. It does host-based prioritization for VOIP, puts mail just underneath bulk traffic, and P2P beyond that (if you have the p2p connmark stuff set). I can completely saturate DSL links with the S518 with this config without appreciable VOIP degradation. I'm using something similar. The missus can talk to her mother (in rural Japan) over IAX while I am using a IPSEC tunnel to work, and doing heavy downloads. Even without an S518, this script works well with external ADSL/cable modems. You may have to play with the upload rate; some cheap ADSL modems will start blocking your upstream traffic beyond as little as 50% of the upstream rate. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two Connected Servers Sound Quailty
One thing I would suggest trying, just from experience, Is the load on the boxes. Unless you have REALLY poor latency, calls do not cut out for just 3-4, but they very well could if the box load is getting very high. Keep a look at top (though not reliable) and the call count when the breakups start happening. Give it a shot :] -bkruse - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 28, 2007 4:51:52 AM (GMT-0800) America/Tijuana Subject: RE: [asterisk-users] Two Connected Servers Sound Quailty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail on Different Server
Hi Steve - Can you elaborate on this, I changed to storing the voicemail via ODBC on MySQL. Each server had it's own local storage, and then MySQL replicated the databases between the sites. This setup was terribly finicky and unstable. It was much worse than the NFS mount. I quickly gave it up. This sounds like it would probably work the best, especially if you have users moving around between offices. What was so finicky and unstable about it? I am not one to quickly give up. I have found that persistence pays off when the idea is sound. Yeah, I thought I had found the silver bullet with MySQL replication (the users do float between offices, so it seemed perfect). There were a number of problems, but in the end it was table corruption as a result of the replication process that made me drop this solution. At the time I set this up, MySQL replication was really designed for one-way replication. Two way replication was possible, but required somewhat unorthodox methods. (Maybe this has changed, I don't know). Configuration is also a little tricky. It's not too bad to set it up between two machines, but 3 machines is more tricky, and 4 is even more tricky, etc, etc. This client had only 3 offices at the time, but I knew they would be expanding. They now have 6. Anyway, after getting everything working, I found that replication would periodically stop after some time. I'd have to re-create the setup, and then replication would work for a time, and then stop again later. This occurred across several different version of MySQL. I suppose I could have fixed this issue with persistence, but unfortunately this was only an annoyance compared to the major issue of data corruption. When replication worked, it was inevitable that after a time the voicemail storage table would experience data corruption. Asterisk did not handle this gracefully at all. It was effectively a total DOS. This also occurred across several versions of MySQL. Sometimes I was able to repair the tables, but usually I couldn't, and the users ended up losing quit a lot of voicemails. I did not have the ability to spend the amount of time I needed to fix the issue, so I scrapped the whole setup. Regular local voicemail storage has been flawless in all installations I've administered. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Audio with SIP to only one provider when switching servers
I snipped all of the previous data, as I'm trying to boil down this problem to its essence... I turned off the firewall for a few seconds, and still got no audio. For those that will be suspicious, the commands were: shorewall stop shorewall clear tested connection, no audio shorewall start I also have a SIPPhone number, which (obviously), connects via SIP. I called that number from the outside, using one of their Access Numbers, and my phone rang and I heard audio in both directions (this with the firewall back on), so SIP definitely works, just not with StanaPhone. Then I connected from another server that I run, which is behind a NAT router. That server is running 1.2.18 (as is the one that isn't working, but is on a public IP). Audio works perfectly with this one. To my knowledge the only difference between them is that the two servers that work are both Red Hat 9, with Asterisk 1.2.18 built from source. The one that fails is CentOS 5.0, with Asterisk 1.2.18 built from source. Here is a dump of the active channel from the NAT'ed server, which _works_: * SIP Call Direction: Incoming Call-ID: [EMAIL PROTECTED] Our Codec Capability: 1822 Non-Codec Capability: 1 Their Codec Capability: 262 Joint Codec Capability: 262 Format ulaw Theoretical Address:204.147.183.18:5060 Received Address: 204.147.183.18:5060 NAT Support:RFC3581 Audio IP: XX.XX.XX.XX (local) Our Tag:as78cfb201 Their Tag: da6aae9eb017f29b6c9de270fb85c352 SIP User agent: Sippy Original uri: sip:204.147.183.55:1024 Caller-ID: XX Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:204.147.183.18;ftag=da6aae9eb017f29b6c9de270fb85c352;lr=on DTMF Mode: rfc2833 SIP Options:(none) The only things edited above are the Audio IP, which is my correct local (before NAT) server address, and my Caller-ID. Everything else is unchanged. Here is the channel with dead audio: * SIP Call Direction: Incoming Call-ID: [EMAIL PROTECTED] Our Codec Capability: 1542 Non-Codec Capability: 1 Their Codec Capability: 262 Joint Codec Capability: 6 Format ulaw Theoretical Address:204.147.183.18:5060 Received Address: 204.147.183.18:5060 NAT Support:RFC3581 Audio IP: XX.XX.XX.XX (local) Our Tag:as45dbcfef Their Tag: 420bab62c5da9eae42686897ae65a385 SIP User agent: Sippy Original uri: sip:204.147.183.55:1024 Caller-ID: XX Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:204.147.183.18;ftag=420bab62c5da9eae42686897ae65a385;lr=on DTMF Mode: rfc2833 SIP Options:(none) The same two fields are edited above, and both were correct. To my eye, these are identical. Both are selecting ulaw, correctly. I'm stumped. I guess that I didn't do any packet tracing, but I'm not sure what the value of that would be given that it's not a firewall problem... Suggestions welcome! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 with skinny on * 1.4.x
Sorry bringing it up again Meanwhile switched to asterisk 1.4.3 on fbsd-6.2 but still no luck getting my 7970G to run via skinny... It registers fine with *: Adding button: 9, 1 Device capability set to '268' asterisk*CLI skinny show devices Name DeviceId IP TypeR NL --- --- - -- ciscoSEP00175A872053 xx.xx.xxx.xx7970Y 1 But on the phone I just see displayed the time and date but no linelabel... My skinny.conf is: [general] bindaddr=xx.xx.xxx.xx ; Address to bind to bindport=2000 ; Port to bind to, default tcp/2000 dateformat=D.M.Y; M,D,Y in any order (5 chars max) keepalive=30 disallow=all allow=all ; see doc/rtp-packetization for framing options [cisco] device=SEP00175A872053 model=7970 nat=1 callerid=Richard Klingler 995 mailbox=995 callwaiting=yes transfer=yes threewaycalling=yes context=klingler linelabel=phonelab line = 995 any ideas left? Using now cmterm-7970_7971-sccp.8-2-2SR1 cheers rick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox/FreePBX
Hi, If this is with an analog card, the 2 ring is normal since the callerid/callername is transmitted on the second ring. Andre Courchesne - Consultant http://www.net-forces.com -- Message: 7 Date: Sat, 28 Apr 2007 13:33:21 -0400 From: [EMAIL PROTECTED] Subject: [asterisk-users] Trixbox/FreePBX To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hello, Installed Trixbox with a digium card and it is taking 2 rings for it to pick up. Any suggestions how to have the system pickup immediately? Thanks, Neal -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070428/7c912f7f/attachment.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.x
A little with skinny debug set to on shows during register: Device SEP00175A872053 is attempting to register Requesting capabilities Buttontemplate requested Adding button: 9, 1 Sending 30006 template to cisco Received SoftKey Template Request Received SoftKeySetReq RECEIVED UNKNOWN MESSAGE TYPE: c Received CapabilitiesRes Adding codec capability '0 (25)' Adding codec capability '4 (4)' Adding codec capability '8 (2)' Adding codec capability '0 (15)' Adding codec capability '0 (16)' Adding codec capability '0 (11)' Adding codec capability '256 (12)' Adding codec capability '256 (12)' Adding codec capability '0 (257)' Device capability set to '268' RECEIVED UNKNOWN MESSAGE TYPE: 49 RECEIVED UNKNOWN MESSAGE TYPE: 49 RECEIVED UNKNOWN MESSAGE TYPE: 4a RECEIVED UNKNOWN MESSAGE TYPE: 9 Received Time/Date Request Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :92 It also show this message when going offhook: RECEIVED UNKNOWN MESSAGE TYPE: 49 Setting ringer mode to '1'. skinny_new: tmp-nativeformats=268 fmt=4 Attempting to Clear display on Skinny [EMAIL PROTECTED] Clearing Display Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :85 Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :11 Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :9a Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :82 Looks to me that chan_skinny doesn't understand many important messages. Any previous 7970G SCCP firmware that might work? cheers rick Richard Klingler schrieb: Sorry bringing it up again Meanwhile switched to asterisk 1.4.3 on fbsd-6.2 but still no luck getting my 7970G to run via skinny... It registers fine with *: Adding button: 9, 1 Device capability set to '268' asterisk*CLI skinny show devices Name DeviceId IP TypeR NL --- --- - -- ciscoSEP00175A872053 xx.xx.xxx.xx7970Y 1 But on the phone I just see displayed the time and date but no linelabel... My skinny.conf is: [general] bindaddr=xx.xx.xxx.xx ; Address to bind to bindport=2000 ; Port to bind to, default tcp/2000 dateformat=D.M.Y; M,D,Y in any order (5 chars max) keepalive=30 disallow=all allow=all ; see doc/rtp-packetization for framing options [cisco] device=SEP00175A872053 model=7970 nat=1 callerid=Richard Klingler 995 mailbox=995 callwaiting=yes transfer=yes threewaycalling=yes context=klingler linelabel=phonelab line = 995 any ideas left? Using now cmterm-7970_7971-sccp.8-2-2SR1 cheers rick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Viable using purchasing sip lines
Hello All, We have been doing Asterisk and CME implementations recently but we almost always exlusively bring in analog lines and or PRI for PSTN access to our systems. I have known about providers providing SIP based lines and SIP trunks to end users for PSTN access. I am curious about the following: - How practical is this? The idea of terminating pstn calls to across the Internet which is an unguarenteed medium concerns me. Even if our access to it is quazi stable T1 data type of access. Do any of you do systems where this is soley the method used for incoming calls from the pstn? If this is done are there things to look for in a SIP provider, as in their presence on the Internet latency ..etc? - What are the major advantages? I know some places provide all you can eat plans which could be seen as a plus and some others provide really low rates. Are there others? - Who are the major players? How are these usually ordered and identified? - Any general tips? Thanks all! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How does Realtime read config files?
Apparently while it was a simple question it was either not a simple answer or no one found it interesting.. I guess i'll give an example: Here is a hard coded queue.conf queue configuration that i would like to put into real time config [CAIS] musicclass = default announce = queue-markq strategy = rrmemory context = queue timeout = 15 retry = 333 weight=0 wrapuptime=90 autofill=yes autopause=yes maxlen = 0 announce-frequency = 90 periodic-announce-frequency=60 ; announce-round-seconds = 10 queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou queue-lessthan = queue-less-than queue-reporthold = queue-reporthold periodic-announce = queue-periodic-announce ;monitor-format = gsm|wav|wav49 ;monitor-type = MixMonitor ;monitor-join = yes joinempty = strict eventwhencalled = yes eventmemberstatus = yes reportholdtime = yes ringinuse = no memberdelay = 1 timeoutrestart = yes Now the issue i've run into is this: the Real time queue is configured with the fallowing mySQL database setup CREATE TABLE queue_table ( name VARCHAR(128) PRIMARY KEY, musiconhold VARCHAR(128), announce VARCHAR(128), context VARCHAR(128), timeout INT(11), monitor_join BOOL, monitor_format VARCHAR(128), queue_youarenext VARCHAR(128), queue_thereare VARCHAR(128), queue_callswaiting VARCHAR(128), queue_holdtime VARCHAR(128), queue_minutes VARCHAR(128), queue_seconds VARCHAR(128), queue_lessthan VARCHAR(128), queue_thankyou VARCHAR(128), queue_reporthold VARCHAR(128), announce_frequency INT(11), announce_round_seconds INT(11), announce_holdtime VARCHAR(128), retry INT(11), wrapuptime INT(11), maxlen INT(11), servicelevel INT(11), strategy VARCHAR(128), joinempty VARCHAR(128), leavewhenempty VARCHAR(128), eventmemberstatus BOOL, eventwhencalled BOOL, reportholdtime BOOL, memberdelay INT(11), weight INT(11), timeoutrestart BOOL ); Now say i would like to use the new feature ringinuse, however this is not a column in the database/table, can i just add a column? will that work... or do i have to wait until an update/fix is made to res_mysql? If you do have to wait until an update is made to res_mysql... why is it designed like this? to me it seems a better design would be to have a database/table setup strictly as KEY VALUE where each column's name is EXACTLY the same as the config option name and the VALUE can be a string with the value... that way when any new config options are added to any of the configuration files you simple just query the database for every key and pull any values found... On 4/26/07, 0xception [EMAIL PROTECTED] wrote: Hi... I just had a real quick and simple question... I have a asterisk implementation setup w/ real time off of a mySQL database for SIP peers and queues, voicemail, agents etc... I after the upgrade to asterisk 1.4.3there are some new configuration features i would like to use. I was wondering if i could just add to the database table a column for the new config option? if this will work or not... For example my queues.conf configuration in the database does not have a column for ringinuse so if i were just just add a column called ringinuse would this work? would this have to be a string for either yes or no or would it be a bool or an integer? thanks for any help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Poor man's High Availability solution
Hi, I'm wondering what the best option to obtain a high availability asterisk server is. I currently use a TE410P (4 x E1) card. I'm thinking of 2 different solutions: - 2 servers configured with Heartbeat + DRBD (drbd mainly for voicemail) and the E1 span plugged to the 2 servers (with a TE410P in each server). - 2 servers configures with Heartbeat + DRBD with the E1 span hooked to an ISDN guard connected to the main server and the backup one. Here comes the real question. Is it technically good to connect an E1 span to 2 cards at the same time (with only one accepting the calls). Since it is possible with BRI cards, i'm wondering if it could be done with PRI. Thanks Laurent ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox/FreePBX
Hi, Thank you all for your replies. The pots lines do not have callerid or fax, turned both of these off as suggested and it now picks up on the first rings. Thanks! Neal On 4/28/07, Andre Courchesne - Consultant [EMAIL PROTECTED] wrote: Hi, If this is with an analog card, the 2 ring is normal since the callerid/callername is transmitted on the second ring. Andre Courchesne - Consultant http://www.net-forces.com -- Message: 7 Date: Sat, 28 Apr 2007 13:33:21 -0400 From: [EMAIL PROTECTED] Subject: [asterisk-users] Trixbox/FreePBX To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hello, Installed Trixbox with a digium card and it is taking 2 rings for it to pick up. Any suggestions how to have the system pickup immediately? Thanks, Neal -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070428/7c912f7f/attachment.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How does Realtime read config files?
Okay i think that real time does work as expected... my issue was actually poor documentation... it seems that everywhere you look call_limit is the configuration option for sip.conf however the REAL option is call-limit not call_limit... the underscore is listed in the initial bug report detailing the need for the call_limit option, it was also in the wiki, and in the main sip.conf file example... On 4/28/07, 0xception [EMAIL PROTECTED] wrote: Apparently while it was a simple question it was either not a simple answer or no one found it interesting.. I guess i'll give an example: Here is a hard coded queue.conf queue configuration that i would like to put into real time config [CAIS] musicclass = default announce = queue-markq strategy = rrmemory context = queue timeout = 15 retry = 333 weight=0 wrapuptime=90 autofill=yes autopause=yes maxlen = 0 announce-frequency = 90 periodic-announce-frequency=60 ; announce-round-seconds = 10 queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou queue-lessthan = queue-less-than queue-reporthold = queue-reporthold periodic-announce = queue-periodic-announce ;monitor-format = gsm|wav|wav49 ;monitor-type = MixMonitor ;monitor-join = yes joinempty = strict eventwhencalled = yes eventmemberstatus = yes reportholdtime = yes ringinuse = no memberdelay = 1 timeoutrestart = yes Now the issue i've run into is this: the Real time queue is configured with the fallowing mySQL database setup CREATE TABLE queue_table ( name VARCHAR(128) PRIMARY KEY, musiconhold VARCHAR(128), announce VARCHAR(128), context VARCHAR(128), timeout INT(11), monitor_join BOOL, monitor_format VARCHAR(128), queue_youarenext VARCHAR(128), queue_thereare VARCHAR(128), queue_callswaiting VARCHAR(128), queue_holdtime VARCHAR(128), queue_minutes VARCHAR(128), queue_seconds VARCHAR(128), queue_lessthan VARCHAR(128), queue_thankyou VARCHAR(128), queue_reporthold VARCHAR(128), announce_frequency INT(11), announce_round_seconds INT(11), announce_holdtime VARCHAR(128), retry INT(11), wrapuptime INT(11), maxlen INT(11), servicelevel INT(11), strategy VARCHAR(128), joinempty VARCHAR(128), leavewhenempty VARCHAR(128), eventmemberstatus BOOL, eventwhencalled BOOL, reportholdtime BOOL, memberdelay INT(11), weight INT(11), timeoutrestart BOOL ); Now say i would like to use the new feature ringinuse, however this is not a column in the database/table, can i just add a column? will that work... or do i have to wait until an update/fix is made to res_mysql? If you do have to wait until an update is made to res_mysql... why is it designed like this? to me it seems a better design would be to have a database/table setup strictly as KEY VALUE where each column's name is EXACTLY the same as the config option name and the VALUE can be a string with the value... that way when any new config options are added to any of the configuration files you simple just query the database for every key and pull any values found... On 4/26/07, 0xception [EMAIL PROTECTED] wrote: Hi... I just had a real quick and simple question... I have a asterisk implementation setup w/ real time off of a mySQL database for SIP peers and queues, voicemail, agents etc... I after the upgrade to asterisk 1.4.3there are some new configuration features i would like to use. I was wondering if i could just add to the database table a column for the new config option? if this will work or not... For example my queues.conf configuration in the database does not have a column for ringinuse so if i were just just add a column called ringinuse would this work? would this have to be a string for either yes or no or would it be a bool or an integer? thanks for any help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Poor man's High Availability solution
Hi Laurent - Is it technically good to connect an E1 span to 2 cards at the same time (with only one accepting the calls). Since it is possible with BRI cards, i'm wondering if it could be done with PRI. Nope. You can use a device like the Redfone fonebridge to convert the PRI to TDMoE. Possible Downside: I've read some reports that say the TDMoE module in asterisk is not so stable. I've heard of a device that acts as a failover for a PRI line so you can plug a PRI into two different devices and have the PRI failover if one device fails. Unfortunately nothing like this is commercially available today. Another option: You can get some backup analog PSTN lines (or an additional PRI), and work with your Telco to do something like a busy-redirect - if the PRI device ever fails, calls go to the PSTN lines. Of course, you'd lose DID capability in this scenario, but at least your calls would go through. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Poor man's High Availability solution
On Sunday 29 April 2007 01:06, Noah Miller wrote: I've heard of a device that acts as a failover for a PRI line so you can plug a PRI into two different devices and have the PRI failover if one device fails. Unfortunately nothing like this is commercially available today. Sounds like the ISDNguard: http://www.junghanns.net/en/ISDNguard_produkt.html HTH -- Med venlig hilsen Sune Kloppenborg Jeppesen kloppenborg.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Poor man's High Availability solution
On Sat, 2007-04-28 at 23:22 +0200, Laurent CARON wrote: Hi, I'm wondering what the best option to obtain a high availability asterisk server is. I currently use a TE410P (4 x E1) card. I'm thinking of 2 different solutions: - 2 servers configured with Heartbeat + DRBD (drbd mainly for voicemail) and the E1 span plugged to the 2 servers (with a TE410P in each server). - 2 servers configures with Heartbeat + DRBD with the E1 span hooked to an ISDN guard connected to the main server and the backup one. Here comes the real question. Is it technically good to connect an E1 span to 2 cards at the same time (with only one accepting the calls). Have a look at the ISDNguard product on the Junghanns.net website: http://www.junghanns.net/en/ISDNguard_produkt.html Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Voicemail on Different Server
How do you handle transfering vmail from one user to another when they're on separate servers? I'm using the single vmail server, mounted NFS partition for this right now. I'd love to be able to have them standalone so they're survivable when the WAN collapses, but I haven't figured out transfer. EKG -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Saturday, April 28, 2007 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail on Different Server Hi Steve - Can you elaborate on this, I changed to storing the voicemail via ODBC on MySQL. Each server had it's own local storage, and then MySQL replicated the databases between the sites. This setup was terribly finicky and unstable. It was much worse than the NFS mount. I quickly gave it up. This sounds like it would probably work the best, especially if you have users moving around between offices. What was so finicky and unstable about it? I am not one to quickly give up. I have found that persistence pays off when the idea is sound. Yeah, I thought I had found the silver bullet with MySQL replication (the users do float between offices, so it seemed perfect). There were a number of problems, but in the end it was table corruption as a result of the replication process that made me drop this solution. At the time I set this up, MySQL replication was really designed for one-way replication. Two way replication was possible, but required somewhat unorthodox methods. (Maybe this has changed, I don't know). Configuration is also a little tricky. It's not too bad to set it up between two machines, but 3 machines is more tricky, and 4 is even more tricky, etc, etc. This client had only 3 offices at the time, but I knew they would be expanding. They now have 6. Anyway, after getting everything working, I found that replication would periodically stop after some time. I'd have to re-create the setup, and then replication would work for a time, and then stop again later. This occurred across several different version of MySQL. I suppose I could have fixed this issue with persistence, but unfortunately this was only an annoyance compared to the major issue of data corruption. When replication worked, it was inevitable that after a time the voicemail storage table would experience data corruption. Asterisk did not handle this gracefully at all. It was effectively a total DOS. This also occurred across several versions of MySQL. Sometimes I was able to repair the tables, but usually I couldn't, and the users ended up losing quit a lot of voicemails. I did not have the ability to spend the amount of time I needed to fix the issue, so I scrapped the whole setup. Regular local voicemail storage has been flawless in all installations I've administered. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to find a codec translation path from ilbc to ulaw
oliver, ugh, it is too obvious... why did it take me so long to figure it out... both phones have to have to negotiate the same codec for audio... as far as I know, * is supposed to do automatic translation and your gateway should be doing translations only on the below codecs. I haven't had that experience yet... one phone may be connected to your * box, but your other phone is *not* connected to *. it is connected to a voip provider... since they don't do any translation other than below. the * connection to webcalldirect must have one of these codecs in the sip.conf for that extension, the extension where webcalldirect is coming in, that is... phoneX - * - webcalldirect - phoneY which one is phone1 and which is phone2? phoneX * -- webcalldirect ---phoneY -| -| - local LAN Internet local LAN some code no codec control no codec control control little or no call quality control the phone connected to * will also select a code that matches up with the caller (webcalldirect)... you have no advantage whether or not * converts the audio to the phone connected to *. you won't get any better reception from webcalldirect because you are not changing that connection. also, I would change iLBC to ilbc, case may make a difference... don't know for sure... perhaps someone else does... hope that is clearer... daveC Codecs G.711 (64 kbps) G.726 (32 kbps) G.729 (8 kbps) G.723 (5.3 6.3 kbps) GSMFR (13.2 kbps) Temporarily unavailable due to technical difficulties. Oliver Brandt wrote: Hi Dave! Thank you very much for replying! what gateway provider are you referring to?doesn't your sip phone webcalldirect (it does not seam to support iLBC directly) connect directly to * as your diagram indicated? Yes, my sipphone ist connected directly to * and also the gateway provider is directly connected to *. My * is on a root server at hosting provider (high bandwith internet connection to the gateway provider) but my phone is connected through DSL with a very limited upstream. For this reason I'd like asterisk to do the codec conversion from iLBC to ulaw. I bett all I have to do is load the codec or/and the codec translator for iLBC to ulaw. But when googleing I only find articles the describe, that * is doing the codec translation automatically. I can't find any information on how to load a codec or the translator manually. I'm probably just using the wrong search string in google... When * starts translators are beeing loaded, but as far as I can see non for iLBC to ulaw. I've put together another test setup with to sip phones to clarify the problem: [phone1] disallow=all allow=iLBC [phone2] disallow=all allow=ulaw When calling from one phone to the other I get the following message: chan_sip.c:2841 sip_call: No audio format found to offer. Cancelling call to phone2 Thank you very much again! Oliver ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_dictate problems
Has no one else experienced the problem I mentioned a few days ago with app_dictate? Or maybe no one is using that app. We're having a problem with choppy audio and failure of the accelerated playback feature which seems to be consistent on a couple of installs, failing with some SIP carriers and working fine with others. MOH and other audio playback features seem to work fine. What's different about app_dictate? -- David Josephson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_CHANCONFIG failedonchannel1:Nosuchdeviceoraddress
On Sat, Apr 28, 2007 at 01:47:15PM +1200, CSB wrote: On Fri, Apr 27, 2007 at 07:11:48AM +1200, CSB wrote: [snip] As suggested earlier I replaced this with: /etc/modprobe.d/zaptel options wctdm opermode=NEWZEALAND honormode=1 boostringer=1 fastringer=1 [snip] dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.17.1 Zaptel Echo Canceller: KB1 wctdm: Unknown parameter `honormode' This is the problem Updated vi /etc/modprobe.d/zaptel options wctdm opermode=NEWZEALAND fxshonormode=1 boostringer=1 fastringer=1 Again, please: rmmod wctdm; modprobe wctdm ; dmesg | tail rmmod wctdm; modprobe wctdm ; dmesg | tail ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wctdm e100: eth1: e100_watchdog: link up, 10Mbps, half-duplex NET: Registered protocol family 10 lo: Disabled Privacy Extensions IPv6 over IPv4 tunneling driver eth0: no IPv6 routers present eth1: no IPv6 routers present Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.17.1 Zaptel Echo Canceller: KB1 Zaptel Transcoder support loaded ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
