Re: [asterisk-users] Re: OpenVox A400P01on thin client?
The Openvox A400P01 is not a full length PCI card. It's a half-length PCI card. You may be referring to the Openvox A1200P (12 port) and that is a full length card. On 5/31/07, Vincent [EMAIL PROTECTED] wrote: On Tue, 29 May 2007 10:23:18 -0300, in gmane.comp.telephony.pbx.asterisk.user Gustavo Cordeiro wrote: No, but I think that you can't install this OpenVox board in this NetStation case, because the card is a full length PCI and the PC case supports only half length PCI cards. Thanks guys for the feedback. I'll check what kind of PCI cards those small form-factor PCs handle. ___ --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] Urgent-- Error while installing app_dtmftotext.
Hi, I am getting the following error after installing SPANDSP along with app_dtmftotext.c file. and while making Asterisk again. Error follows:: *** [EMAIL PROTECTED] asterisk-1.4.1]# make Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. [CC] app_dtmftotext.c - app_dtmftotext.o app_dtmftotext.c:102: warning: data definition has no type or storage class app_dtmftotext.c:102: warning: type defaults to int in declaration of STANDARD_LOCAL_USER app_dtmftotext.c:104: warning: data definition has no type or storage class app_dtmftotext.c:104: warning: type defaults to int in declaration of LOCAL_USER_DECL app_dtmftotext.c: In function festival_exec: app_dtmftotext.c:339: warning: implicit declaration of function ast_load app_dtmftotext.c:339: warning: assignment makes pointer from integer without a cast app_dtmftotext.c:345: warning: assignment discards qualifiers from pointer target type app_dtmftotext.c:349: warning: assignment discards qualifiers from pointer target type app_dtmftotext.c:357: warning: assignment discards qualifiers from pointer target type app_dtmftotext.c:365: warning: assignment discards qualifiers from pointer target type app_dtmftotext.c:369: warning: assignment discards qualifiers from pointer target type app_dtmftotext.c:382: warning: implicit declaration of function LOCAL_USER_ADDapp_dtmftotext.c:396: warning: implicit declaration of function __gethostbyname__is__not__reentrant__use__ast_gethostbyname__instead__ app_dtmftotext.c:396: warning: assignment makes pointer from integer without a cast app_dtmftotext.c:535: warning: implicit declaration of function LOCAL_USER_REMOVE app_dtmftotext.c: At top level: app_dtmftotext.c:1031: warning: no previous prototype for unload_module app_dtmftotext.c: In function unload_module: app_dtmftotext.c:1032: error: STANDARD_HANGUP_LOCALUSERS undeclared (first use in this function) app_dtmftotext.c:1032: error: (Each undeclared identifier is reported only once app_dtmftotext.c:1032: error: for each function it appears in.) app_dtmftotext.c: At top level: app_dtmftotext.c:1037: warning: no previous prototype for load_module app_dtmftotext.c:1042: warning: no previous prototype for description app_dtmftotext.c:1047: warning: no previous prototype for usecount app_dtmftotext.c: In function usecount: app_dtmftotext.c:1050: warning: implicit declaration of function STANDARD_USECOUNT app_dtmftotext.c: At top level: app_dtmftotext.c:1055: warning: function declaration isnt a prototype make[1]: *** [app_dtmftotext.o] Error 1 make: *** [apps] Error 2 You have new mail in /var/spool/mail/root [EMAIL PROTECTED] asterisk-1.4.1]# *** Please help me in this issue. it's very urgent. Regards K.Rajesh. _ Tried the new MSN Messenger? Its cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in ___ --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] False ring problem
Well, you r right. This was the carrier`s fault. Its been removed on our request and now we r okay. thanx to all. On 5/31/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Rizwan Hisham wrote: Nobody is using r option anywhere in my dialplan, thats 4 sure. And im also not using any PSTN line to connect to outside world. my system is based on voip only. SIP PHONE--ASTERISK--CARRIER-OUT My only idea is that the carrier might be using the r option. If they are then you should switch carriers. Also callprogress=yes might cause the problem you are experiencing, but I doubt this is the case. ___ --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 -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --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] moh backround?
On Thursday 31 May 2007, Alex Balashov wrote: Sadly, I don't think this is possible. The only sense in which Background() plays anything in the background is that it allows the caller to interrupt the playback with extension input / DTMF, instead of that input polling being deferred until the end of the playback. You would have to use a sound mixing program and reduce the volume on the hold music, and then superimpose the sounds you want on it. Thank you. regards t. -- knowledgeTools® ... managing complexity. -- knowledgeTools International GmbH Wallstraße 15 / 15 a 10179 Berlin Fon: +49 30 726 169 20 Fax: +49 30 726 169 249 [EMAIL PROTECTED] www.knowledgetools.de Sitz Berlin, AG Berlin-Charlottenburg, HRB 86378 Geschäftsführer: Oliver Seyboldt, Reinhard Kunz -- This eMail communication (and any attachment/s) may contain confidential or privileged information and is intended only for the individual(s) or entity named above and to others who have been specifically authorized to receive it. If you are not the intended recipient, please do not read, copy, use or disclose the contents of this communication to others. Please notify the sender that you have received this e-mail in error by reply e-mail, and delete the e-mail subsequently. Thank you. ___ --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] CARD FOR inband signal
On Fri, Jun 01, 2007 at 11:35:24AM +0800, clive.chan(Alpha Trilogies Networks) wrote: Hi all, I wish to use analog interface card for the inband capturing media and use the Asterisk Open Source as a core software. I have tried the Sangoma card, and Digium card, and found that the inabnd can't capture from some PBX system. Why ?? What do you mean by inband signalling? Could you please be more specific? Any specific type of inband signal? Or any other hardware card has such features in order to work with all the inband capturing? Or may be some of your has such experience, and I am welcome to share with. -- 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
RE: [asterisk-users] Audio going blank for a few seconds and thencomes back. What could be the reason?
I think this is more related to the PRI, we've been seeing this for a few weeks now, and our environment is bridged PRI-PRI on the same board, Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria Sent: 10 May 2007 01:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Audio going blank for a few seconds and thencomes back. What could be the reason? I have Grandstream and Aastra phones. It happens on both of them. The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk___ --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] moh backround?
On Friday 01 June 2007, Dave Bour wrote: Using the idea of a week ago for moh, what about using a conference bridge for it? Dave Bour What article are you referring to? regards t. -- knowledgeTools® ... managing complexity. -- knowledgeTools International GmbH Wallstraße 15 / 15 a 10179 Berlin Fon: +49 30 726 169 20 Fax: +49 30 726 169 249 [EMAIL PROTECTED] www.knowledgetools.de Sitz Berlin, AG Berlin-Charlottenburg, HRB 86378 Geschäftsführer: Oliver Seyboldt, Reinhard Kunz -- This eMail communication (and any attachment/s) may contain confidential or privileged information and is intended only for the individual(s) or entity named above and to others who have been specifically authorized to receive it. If you are not the intended recipient, please do not read, copy, use or disclose the contents of this communication to others. Please notify the sender that you have received this e-mail in error by reply e-mail, and delete the e-mail subsequently. Thank you. ___ --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] Net2Phone Multiple SIP Trunk Not Working
You just have a 1 call limit on your account on net2phone side . Making 10 trunk wont let you make 10 account its restriction on your account not ip . Just change your provider . On 01/06/07, Salah Eddine ELMRABET [EMAIL PROTECTED] wrote: Hi, Any help regarding Net2Phone poblem? BR On 6/1/07, Andrew Furey [EMAIL PROTECTED] wrote: On 01/06/07, Salah Eddine ELMRABET [EMAIL PROTECTED] wrote: I'm sorry that's because I didn't get a visibility of ny post, I though that was a network problem (as I cannot see my post on the mailing list) You never do with mailing lists on Gmail, I presume it hides it based on the message ID (since you already have a copy). Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --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] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO!
Well freepbx runs fine with asterisk 1.4.4 ( atleast for me ) i think some changes was introduced in 1.4 ( 1.4.4 ?) for some backward compatibility... like show channels now work in 1.4.4 instead of core show channels but it gives a notice that 'show channels' is deprecated bla bla .Freepbx works completely fine with asterisk 1.4 for me . On 31/05/07, shadowym [EMAIL PROTECTED] wrote: If anything this should motivate the FreePBX developers a bit more. -Original Message- From: Jared Smith [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 30, 2007 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO! On 5/30/07, BSumrall [EMAIL PROTECTED] wrote: AMP does not support 1.4 and will not until AMP 2.3 is released! I'm sorry to hear you think our decision (I say our, as I was at the Asterisk Developers' Conference where the decision was made) will kill the AMP project. Personally, I don't think the situation is as dire as you say. I'm quite sure the AMP developers will step up to the plate and support Asterisk 1.4 in due time. When that will be I can't say, as I'm not active in the AMP community. I can't image it would take that long to move over to Asterisk 1.4, as the dialplan changes aren't *that* extensive between 1.2 and 1.4. (Obviously any code that ties into the internal C APIs of Asterisk will take longer to port.) Bet you guys didn't think about that one! Actually, we did. As a matter of fact, I was *very* vocal at the conference in stating that we needed to give users, integrators, and projects like AMP a substantial warning before putting Asterisk 1.2 in security maintenance mode, as they need time to react. At the same time, I don't think anyone should expect the Asterisk developers to base all their decisions completely on the timetables of outside projects (like AMP). There is a plethora of projects and programs out there that tie into Asterisk, and if we as developers waited for every single one to move over to Asterisk 1.4, we'd never accomplish anything. There's simply a finite set of resources (developers and bug marshalls in this case), and a decision had to be made on how best to use those resources. Personally, I think it would be great if there were more communication between the outside projects and the Asterisk developers, so that there isn't so much animosity when decisions like this are made. In short, the decision is probably going to cause some short-term discomfort for some people, but I truly believe it's a good decision for the long-term health and sanity of the Asterisk developers and Asterisk community in general. No, we're not trying to kill off AMP or any other outside project -- we're trying to make Asterisk (and by extension, anything that uses or adds on to Asterisk) as great as possible. -Jared ___ --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] how can qualify=yes trigger some external event?
Hi all, The option qualify=yes allows Asterisk to check if it can reach the peer. If the device does not answer within the time-out period, Asterisk considers the device off-line for future calls. Is it possible to use this feature to trigger some external event, in case of failed reply from the peer that is tried to be reached? How can that be done? Regards, Ricardo. ___ --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] Net2Phone Multiple SIP Trunk Not Working
Hi, I'm using 10 different accounts, once the first trunk is on use the second one cannot be used even if the result of chanisavail refer to the second one. Also when I choose the second trunk as only route it doesn't work. Regards, On 6/1/07, Jaswinder Singh [EMAIL PROTECTED] wrote: You just have a 1 call limit on your account on net2phone side . Making 10 trunk wont let you make 10 account its restriction on your account not ip . Just change your provider . On 01/06/07, Salah Eddine ELMRABET [EMAIL PROTECTED] wrote: Hi, Any help regarding Net2Phone poblem? BR On 6/1/07, Andrew Furey [EMAIL PROTECTED] wrote: On 01/06/07, Salah Eddine ELMRABET [EMAIL PROTECTED] wrote: I'm sorry that's because I didn't get a visibility of ny post, I though that was a network problem (as I cannot see my post on the mailing list) You never do with mailing lists on Gmail, I presume it hides it based on the message ID (since you already have a copy). Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --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 ___ --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] ZAP inbound/outbound connection taking too long
On Fri, 1 Jun 2007, Gavin Henry wrote: Dear all, I think this is common, or at least how it is supposed to be, but whening dialing over a ZAP channel, it's taking around 5~ seconds to ring on the over end, likewise inbound. This is just with a normal Dial command. It's normal for an analogue Zap channel. Asterisk has to sieze the line (after a basic check to make sure the channel is free), that may entail a delay of a second or so while it makes sure there there is a dial-tone (actually, I'm not sure it waits for a dial-tone), then it sends the digits out via DTMF - that might take a second or 2 for a long number - then it's up to the PSTN switch at the other end to connect the call - depending on the technology, this might take several seconds. What you can do is connect to asterisk (asterisk -r), set verbose , then initiate a dial and you'll see the dialplan progress and you can work out yourself where the longest part of the delay is... Inbound ought to be answered as soon as asterisk hears the ringing signal - but this might be one whole ring time from the ring starting, depending on how caller-id is being handled in your country, again, monitor it by looking at the output on the console, and by connecting an existing analogue phone in paralel with the incoming Zap line. Gordon ___ --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] Context documentation for the newbie!
Bsumrall, Take a look on this document, http://bef.eventphone.de/a/Ast.%20C.%20I._files/ast-ci-draft1.pdf /Mats On 6/1/07, C F [EMAIL PROTECTED] wrote: I can give the following example, let me know if it helps. Mr 1 has a child Mr 10 and another child Mr 11, now Mr 10 has Mr 100 and Mr 11 has Mr 111. Mr 10 adopts Mr 111. Also Mr 88 adopts Mr 10. Which brings us to the family tree, if you are a child of one, you are a grandchild of that ones parent, and as such included in that tree. Now one of the children could be adopted by some other parent as well, which makes that child a child of another parent hence a grandchild of that parents parent. Subistute child and adopt for include =, and Mr for context so you got: [1] include = 10 include = 11 [10] include = 100 include = 111 [11] include = 111 [88] include = 10 Within each context you got the instruction code, which is an extension (exten) prioritized with numbers (or n for next number). The instructions are executed one after the other, unless a jump is encountered. Each extension is a pointer within that context that starts the instruction set. In Asterisk one starts in a context, when an extension is called (by dialing, or s when the extension number wasn't given) Asterisk looks for that extension in that context, if it can't find it there it searches in that contexts family tree, if still no match it searches in default context, if still no match it searches for the i extension in the same order, if still no match then 404 is given. Hope this helps. On 5/31/07, BSumrall [EMAIL PROTECTED] wrote: Does anyone know where there is better documentation on understanding context relations and priorities with examples? http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction Does tell me anything other than they point to each other. Not how or who comes first or even how to get them to work with each other! ___ --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] High Port Count ATA
You can add their gateway blade to convert to voip via ethernet, but it only does mgcp. How about doing GR303 to an access navigator with channel banks hanging off that? Pricey but carrier class gear and scales WAY up. Could also do Adtran total Access concentrator (4303?) feeding their total Access 1500 with TR08 would be more dense and possibly cost less. Best way is also going to be determined by how many calls up at one time. Going with one of the 48port sip gateways may be ok if locally peered with the Asterisk server. On May 31, 2007, at 5:49 PM, Douglas Garstang wrote: Cory, I’m not quite clear on that. Do these channels banks have an IP uplink port so that each FXS port can SIP register to asterisk? Doug. From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Thursday, May 31, 2007 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] High Port Count ATA Channel banks would work. Rhino works well, or if you need more chassis density, try the Carrier Access ADIT600 configured with FXS blades. Cory J Andrews From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Thursday, May 31, 2007 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] High Port Count ATA I’m trying to find a high port count ATA device. We have a lot (up to 110) analog devices that we need to bridge to IP. Rather than go out and buy 110 ATA’s, it would make more sense to buy a single chassis type device with some number of ports and blades. Anyone know if such a device exists? Thanks, 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
[asterisk-users] ZAP inbound/outbound connection taking too long
Dear all, I think this is common, or at least how it is supposed to be, but whening dialing over a ZAP channel, it's taking around 5~ seconds to ring on the over end, likewise inbound. This is just with a normal Dial command. Are there any ways to tweak this? Thanks, Gavin. ___ --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] Audio going blank for a few seconds and thencomes back. What could be the reason?
There are some remote extensions connected on this system, and calling long distance is purely on voip. These remote extensions also face the same thing, i.e. audio going blank for a few seconds, when dialing long distance. So in this case, no PRI is involved. Its either the server, or the network. Now I don't know how to find out what is it and why? On 6/1/07, Steve Hanselman [EMAIL PROTECTED] wrote: I think this is more related to the PRI, we've been seeing this for a few weeks now, and our environment is bridged PRI-PRI on the same board. 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
[asterisk-users] OT Slightly:
Interesting article in this months SB http://www.strategy-business.com/press/enewsarticle/enews053107?pg=0 Written by Nicholas Carr - The Ignorance of Crowds The open source model can play an important role in innovation, but know its limitations. At first pass I dissed it and was about to write back to Art Kleiner the editor about how BAH should stick to what it knows and was about to provide references on the Asterisk development as a shining example of Open Source at it's best..but when you read it the second or third time on the 3rd and 4th page it starts to get interesting. Maybe the implementation Digium/Asterisk has struck is a perfect example of crowd development but with centralized control. Anyway I'm throwing it out there for what it's worth and hope it's of interest. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --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] how can qualify=yes trigger some external event?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Friday, June 01, 2007 6:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] how can qualify=yes trigger some external event? Hi all, The option qualify=yes allows Asterisk to check if it can reach the peer. If the device does not answer within the time-out period, Asterisk considers the device off-line for future calls. Is it possible to use this feature to trigger some external event, in case of failed reply from the peer that is tried to be reached? How can that be done? Regards, Ricardo. Hi, Ricardo. Currently there is no way to do this in a pure configuration-only sort of way. However, if you're even moderately adept at C a cursory glance through chan_sip.c will show that it would be quite straightforward to modify the code in order to allow this. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] reset Polycom phones remotely
Are you able to access the phone via a web browser? And did asterisk register the phone? If both are true and you set the always reboot flag to 1, then rebooted the phone by hand, there shouldn't be anything standing in the way. Rob Stephen Bosch wrote: Rob Schall wrote: Correct. Once this is set to 1, then it will reboot regardless. I've been using this effect for over a year. Hmn -- just tried this. It doesn't seem to be working... -Stephen- ___ --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] Audio going blank for a few seconds andthencomes back. What could be the reason?
You can use tcpdump or ethereal (wireshark now) to capture the stream and then see if there was loss during the call, just leave a capture going then get your users to mark out the time at which they encountered the silence, compare this to the server time (e.g. their watch to the server) to get a time difference, then figure out what time you need to look at in the trace. Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria Sent: 01 June 2007 13:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason? There are some remote extensions connected on this system, and calling long distance is purely on voip. These remote extensions also face the same thing, i.e. audio going blank for a few seconds, when dialing long distance. So in this case, no PRI is involved. Its either the server, or the network. Now I don't know how to find out what is it and why? On 6/1/07, Steve Hanselman [EMAIL PROTECTED] wrote: I think this is more related to the PRI, we've been seeing this for a few weeks now, and our environment is bridged PRI-PRI on the same board, Steve The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk___ --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] Thank you Asterisk mailing list!
I was very happy to hearing your story Brad. A couple of times almost the same thing happened with me. Problems with NAT, module compilations, that I could solve without sending a single question to the list: Just searching for its arquive. As a reflection, all the Free Software/Open Source world became huge as it is today, due to proactive work of its members. Answering questions and working on code without imediate financial rewards. Thinking this way, I invite those who think about the open source communities just as a zero price and its mailing lists as a space to wait passively for answers, to rethink its own ideas. Before asking for something and adding trash to communities mailing lists, DO A WORK OF RESEARCH and then, send to the list what you made to solve (the solutions), even if you don't need the help of the list for that issue anymore. Thanks for the oportunity to talk about this, Brad. Rgds, Ricardo Martins. BSumrall escreveu: After 3 days of crunch and a whole lot of reading. This mailing list, and this mailing list alone has led me to the solutions to my answers. I am officially past the learning curve and have a blunt understanding of what is going on. Any questions now will be detailed and with knowledge. You can lead a horse to water, but you cannot make him drink! I gulped! And now with this dialplan thing and hardcore internal troubleshooting, I have a definite course of action. I have worked with asterisk for years but have always gaffed off the dialplan to peers and contractors and finally decided it was time to learn this puppy! Thank you for helping me get through the curve! Brad ___ --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
max verbose level [was: Re: [asterisk-users] ZAP inbound/outbound connection taking too long]
Unrelated issue: On Fri, Jun 01, 2007 at 12:03:27PM +0100, Gordon Henderson wrote: On Fri, 1 Jun 2007, Gavin Henry wrote: What you can do is connect to asterisk (asterisk -r), set verbose , Any point in verbose level over 4 ? -- 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
Re: max verbose level [was: Re: [asterisk-users] ZAP inbound/outbound connection taking too long]
On Fri, 1 Jun 2007, Tzafrir Cohen wrote: Unrelated issue: On Fri, Jun 01, 2007 at 12:03:27PM +0100, Gordon Henderson wrote: On Fri, 1 Jun 2007, Gavin Henry wrote: What you can do is connect to asterisk (asterisk -r), set verbose , Any point in verbose level over 4 ? Probably not - I just seem to have gotten into the habit of typing that :) Gordon ___ --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] RF to IP bridge
Curt, Have a look here, www.app-rpt.qrvc.com www.qrvc.com/radiocards.html John Treble Ottawa, Ontario -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: May 31, 2007 7:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] RF to IP bridge Half duplex is not an issue. Basically the idea is radio over IP. I don't want to change the fact that we are using radios. For example, on an enterprise level I'm going to be working with a crew to set this up for our Avaya system. It is basically for emergency communications. Say the fire chief is out of town and something major happens. We would like for him to be able to call in and hear and interact with the squad on site via the radio network from the PSTN or even a cell phone. With http://www.twistpair.com/ this is completely possible but that only integrates with Avaya or Cisco Call Manager at this time. Not a problem as we run Avaya on an Enterprise level but I'm looking for free or cheap alternatives. Another example and more towards what I am looking at. As a RACES (Radio Amateur Civil Emergency Service) member I would like to have a crash cart that would allow instant ability for communications on a range of mediums. GSM cards, EVDO, WIFI, and radio communications all from a small box that can be very mobile and run on something like a gel cell batteries. The ability to bridge between the two would be very useful in cases of disperse conditions where every RACES member could be offering communications to victims outside of net repeaters or have another medium to get back into the tactical net rather than having to utilize repeaters out of the range of the net control. We have internet controlled repeaters and utilize VoIP on a lot of them but we are looking for something that can be small, very mobile and offer other services other than just radio communications. And just FYI the ~$200/channel is for the above named software that does just what I'm explaining. Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Coccimiglio Sent: Thursday, May 31, 2007 6:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RF to IP bridge Per Jessen wrote: Radio-amateurs have done phone-patching for decades (where allowed) - there must be someone who can point you in the direction of an easy solution. /Per Jessen, Zürich The BIG problem here is that most Radio Amateur software and hardware operate in a half-duplex manner. I don't think that would be what you want. If half-duplex is ok then most radio makers (Icom, Motorola, etc.) have complete turn-key solutions. If you want it cheap then your will have to build it yourself. I don't see $200/channel happening in either case for VHF/UHF. Please share more info and maybe I can help. Mark C ( N3WHX ) [EMAIL PROTECTED] sip:[EMAIL PROTECTED] (VoIP) ___ --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] Audio going blank for a few seconds andthencomes back. What could be the reason?
We have the same problem with our system. Unless you have a solid (not just high speed) connection between the 2 parties, you're going to get silence a few times during the call. We had set up a user on a business comcast high-speed, thinking that would be more than enough. Turned out though, with most high speed solutions, there is some limited packet loss and its just to be expected. You internet browsers, etc, would normally just re-request the packet and move on, but with a stream, you're out of luck. The only real solution is to have a dedicated T1 or mpls connection or something like that for perfect quality. We have solid connections between our offices and haven't had a problem yet. Steve Hanselman wrote: You can use tcpdump or ethereal (wireshark now) to capture the stream and then see if there was loss during the call, just leave a capture going then get your users to mark out the time at which they encountered the silence, compare this to the server time (e.g. their watch to the server) to get a time difference, then figure out what time you need to look at in the trace. Steve *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Zeeshan Zakaria *Sent:* 01 June 2007 13:02 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason? There are some remote extensions connected on this system, and calling long distance is purely on voip. These remote extensions also face the same thing, i.e. audio going blank for a few seconds, when dialing long distance. So in this case, no PRI is involved. Its either the server, or the network. Now I don't know how to find out what is it and why? On 6/1/07, *Steve Hanselman* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I think this is more related to the PRI, we've been seeing this for a few weeks now, and our environment is bridged PRI-PRI on the same board. Steve The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendataco.uk ___ --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] High Port Count ATA
i have deployed the audiocodes mp-124? with 14 lines active lines and it ugly to configure, but works well once setup. They do make it easy if you have a set of contiguous number to apply to the ports in order though. On 5/31/07, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, May 31, 2007 at 01:22:06PM -0700, Douglas Garstang wrote: I'm trying to find a high port count ATA device. We have a lot (up to 110) analog devices that we need to bridge to IP. Rather than go out and buy 110 ATA's, it would make more sense to buy a single chassis type device with some number of ports and blades. Anyone know if such a device exists? Sure; lots of them, in port counts up to 48. They're not cheap. Expect to pay about $105 a port or so, on average. Start here: http://www.voipsupply.com/index.php?cPath=94_286 I was looking at the Audiocodes, about 6 months ago when I was prepping a project, but I haven't had hands-on. Iv you *really* need a lot of ports, check out the Vegastream, but, again, no testimony here. Keep in mind though, that it can be a good thing to have a warm spare, so high port counts aren't *always* the best answer. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 ___ --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] Audio going blank for a few seconds andthencomes back. What could be the reason?
On Friday 01 June 2007 9:24 am, Rob Schall wrote: comcast high-speed, thinking that would be more than enough. Turned out though, with most high speed solutions, there is some limited packet loss and its just to be expected. You internet browsers, etc, would Limited packet loss != **EIGHT SECONDS** of network breakage. Jitter buffers and PLC takes care of most normal network indiscretions, but period dropouts of that big of a time aren't normal and indicate a bigger issue, either with the hardware or the link itself. normally just re-request the packet and move on, but with a stream, you're out of luck. The only real solution is to have a dedicated T1 or mpls connection or something like that for perfect quality. We have solid connections between our offices and haven't had a problem yet. I have numerous installations using standard telco (Bell Canada and Telus) DSL, and at least one on Rogers cable here in Ontario. No real problems. The odd problem if the pipe gets saturated but careful design and monitoring can take care of most of these problems. I agree with Mr. Hanselman; get a packet logger on the link and see what's really going on. Until that's done, everything here is just speculation. I have seen bugs in the IAX2 and SIP jitter buffers on Asterisk which cause dropouts like this, and I'd like to see what's actually going on before pointing any fingers. -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] SIP NAT ...
So I thought I had SIP and NAT cracked a long time ago, but something's just happened that's sort of upset the cart )-: I have an * box behind a NAT firewall. Nothing unusual there, this is something I've done many times - sip.conf has the correct nat= localnet= externip= settings, the router has ports 5060-5069 and 1-2 forwarded to the internal IP address of the * box. (and 4569 for IAX, but we're just using SIP here) The * server has a few internal (LAN) and external SIP phones, but also has 2 SIP connections to an external PSTN provider. I don't know what this is as I don't have any control or access to it, but both go to the same IP address with different account details (username/passwords) Both these SIP - external PSTN provider connections register OK on the * box, and outgoing calls placed over either connection works perfectly. Outgoing callerId (set by the external provider) works as expected. ) I have dialling prefixes for each 'line', nothing special there, that side of it all works as expected. The problem is that only the last one in the sip.conf file actually accepts incoming calls when dialled from the PSTN side. (They have different PSTN phone numbers) If I swap their entries over in the sip.conf file, then the other one takes the calls. When dialling the first number, nothing seems to get through to the * box at all - nothing on the console in verbose mode, nothing in the log-file. The 2 SIP account setups are otherwise identical (generated by a web interface), just the usenrname password differing, and the account name. Anyone seen this before? I'm wondering if it's an issue with the rotuter (Draytek 2800 ADSL), or is there an issue with 2 SIP channels to the same external IP address (port clash?) I've tried with without bindport= settings in the sip.conf file too - doesn't make any difference. Any clues appreciated! Gordon ___ --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] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
Matthew J. Roth wrote: Recently, we were pushing our server to almost full CPU utilization. Since we've observed that Asterisk is CPU bound, we upgraded our server from a PowerEdge 6850 with four single-core Intel Xeon CPUs running at 3.16GHz, to a PowerEdge 6850 with 4 dual-core Intel Xeon CPUs running at 3.00GHz. The software installed is identical and a kernel build benchmark yielded promising results. The new dual-core server ran roughly 80% faster, which is about what we expected. As far as Asterisk is concerned, at low call volumes the dual-core server outperforms the single-core server at a similar rate. Outperforms in what sense? I'm working on a follow-up post that will demonstrate this with some benchmarks for a small number of calls in various scenarios on each machine. However, to our surprise as the number of concurrent calls increases, the performance gains begin to flatten out. In fact, it seems that somewhere between 200 and 300 calls, the two servers start to exhibit similar idle times despite one of them having twice as many cores. What do you mean by idle here? ___ --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
OT: The Ignorance of Crowds (was: [asterisk-users] OT Slightly: )
I see what Dean means about how Digium/Asterisk might have struck a balance between the cathedral and the bazaar antipodes of the SW development world. Nicholas Carr's The Ignorance of Crowds finally states his politics when it says When you move from the bazaar to the cathedral, it’s best to leave your democratic ideals behind. But treating open/closed source/projects as a pure dichotomy of two extremes of openness is a purely ideological exercise: and one that favors the cathedral, the very institution of ideology rather than practice. There are many degrees of openness, even just in the categories of the source code and of the project management. There are degrees of openness in the readability, writeability and executeability in each of those categories, to extend a metaphor. And there are other abilities, like redistribution, documentation, training, etc, which can be open to varying degrees. And any project can mix practically any openness degree in practically each of those abilities, for a vast combinatoric range. And calling the bazaar democracy is to misunderstand, and probably treat with contempt, both democracy and the *anarchy* of the market. Even the article's example that Dean highlighted, Wikipedia, shows no real democracy, even the pure Athenian version that few Americans (except maybe some Californians) would recognize. Without actual rule by all of its contributors and readers, but rather primary rule by many policies determined and (often) enforced by people selected by autocrats (however benevolent), it's no democracy, but rather a collegiocracy or something else with a new name. Digium/Asterisk is an interesting example. For example, the community has so far accepted the proprietary ownership of code contributed to Digium, but a tension in source code openness lies in that degree in that category. The recent decision to stop new development of 1.2 in favor of 1.4 has just begun to enter the community consciousness, but the state of 1.4 when the 1.2 deadline comes will probably demonstrate limits of the project's openness to at least some committed 1.2 users/developers. Digium's Asterisk trademark hasn't yet become an issue, AFAIK, but a confusingly named fork, or just competing app from a different codebase with a very similar name could make all the current Aster* names into precedent damaging to the trademark, if not the mark itself. Digium is a corporation: an autocracy, not a democracy. It offers no data to judge democracy in its cathedral ruling its bazaar. And there are no deductively identical but for one versions of Digium run instead as a democracy to which to directly compare. Cathedral/bazaar is not a binary choice. They're more like antitheses that projects combine into a synthesized community model somewhere in the sphere of control combinations. It's too early to judge Digium's Asterisk success, let alone use it as a benchmark to calibrate cathedral/bazaar combinations. At least we have some terms in which we can model these complex behaviors and try to compare them. I don't think either the bazaar or the cathedral is in any way limited by, or alien to, democratic ideals. A much more wise politics comes from Yogi Berra, who said there is no difference between theory and practice - in theory. Let's keep trying the best way of running each job, and judge from the results when we've got examples of each. We can call them names when they've demonstrated what precedents they're actually like, and who likes them. What do you think? On Fri, 2007-06-01 at 05:42 -0700, [EMAIL PROTECTED] wrote: Date: Fri, 1 Jun 2007 08:42:48 -0400 From: Dean Collins [EMAIL PROTECTED] Subject: [asterisk-users] OT Slightly: To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Interesting article in this months SB http://www.strategy-business.com/press/enewsarticle/enews053107?pg=0 Written by Nicholas Carr - The Ignorance of Crowds The open source model can play an important role in innovation, but know its limitations. At first pass I dissed it and was about to write back to Art Kleiner the editor about how BAH should stick to what it knows and was about to provide references on the Asterisk development as a shining example of Open Source at it's best..but when you read it the second or third time on the 3rd and 4th page it starts to get interesting. Maybe the implementation Digium/Asterisk has struck is a perfect example of crowd development but with centralized control. Anyway I'm throwing it out there for what it's worth and hope it's of interest. Regards, Dean Collins -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
Sean M. Pappalardo wrote: Hi there. Just curious if you've checked out Linux clustering software such as OpenSSI ( http://www.openssi.org/ ) and run Asterisk on it? It features a multi-threaded cluster-aware shell (and custom kernel) that will automatically cluster-ize any regular Linux executable (such as the main Asterisk process.) If it works as advertised, it should just be a matter of adding boxes to the cluster to speed up processing. OpenSSI can't (at the moment) migrate threads between compute nodes. It can migrate separate processes, but doesn't Asterisk use threads? ___ --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] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks
Matthew J. Roth wrote: This post contains the benchmarks for Asterisk at low call volumes on similar single and dual-core servers. I'd appreciate it greatly if you took the time to read and comment on it. For me all these numbers look too small to be useful for benchmarking. ___ --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] G729 client and server Side
Hi iam using G729 at server side and same iam using eyebeam with g729 at client side still its take transcoding CPU process or its pass through ram ___ --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] Audio going blank for a few seconds andthencomesback. What could be the reason?
There seem to be two threads here that mention multi-second loss with the common part being a PRI, certainly for my situation it's purely PRI as the asterisk box sits in between the telco and another PRI enabled PBX and the calls are bridged between the two. There is no network traffic involved in this case. Not sure where to go with mine though, the load average is nice and low, I don't see any missed interrupts and it's only started happening in the last few weeks since an asterisk upgrade. Latest FC6 kernel, latest yum'd asterisk, zaptel etc Not sure whether it's worth pulling a SVN version down and building that, the only issue is I can't currently reproduce this on demand. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: 01 June 2007 14:36 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Audio going blank for a few seconds andthencomesback. What could be the reason? On Friday 01 June 2007 9:24 am, Rob Schall wrote: comcast high-speed, thinking that would be more than enough. Turned out though, with most high speed solutions, there is some limited packet loss and its just to be expected. You internet browsers, etc, would Limited packet loss != **EIGHT SECONDS** of network breakage. Jitter buffers and PLC takes care of most normal network indiscretions, but period dropouts of that big of a time aren't normal and indicate a bigger issue, either with the hardware or the link itself. normally just re-request the packet and move on, but with a stream, you're out of luck. The only real solution is to have a dedicated T1 or mpls connection or something like that for perfect quality. We have solid connections between our offices and haven't had a problem yet. I have numerous installations using standard telco (Bell Canada and Telus) DSL, and at least one on Rogers cable here in Ontario. No real problems. The odd problem if the pipe gets saturated but careful design and monitoring can take care of most of these problems. I agree with Mr. Hanselman; get a packet logger on the link and see what's really going on. Until that's done, everything here is just speculation. I have seen bugs in the IAX2 and SIP jitter buffers on Asterisk which cause dropouts like this, and I'd like to see what's actually going on before pointing any fingers. -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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --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] False ring problem
In my opinion, any carrier that adds r to a Dial line without a VERY, VERY good reason is not a carrier that I want to use. Using r is a classic newbie problem. It indicates a serious lack of understanding about Asterisk. Rizwan Hisham wrote: Well, you r right. This was the carrier`s fault. Its been removed on our request and now we r okay. thanx to all. On 5/31/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Rizwan Hisham wrote: Nobody is using r option anywhere in my dialplan, thats 4 sure. And im also not using any PSTN line to connect to outside world. my system is based on voip only. SIP PHONE--ASTERISK--CARRIER-OUT My only idea is that the carrier might be using the r option. If they are then you should switch carriers. Also callprogress=yes might cause the problem you are experiencing, but I doubt this is the case. ___ --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] Asteris et winsip
Does anyone tried the Winsip sotware to test Asterisk? _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE___ --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] SIP NAT ...
Gordon Henderson wrote: So I thought I had SIP and NAT cracked a long time ago, but something's just happened that's sort of upset the cart )-: I have an * box behind a NAT firewall. Nothing unusual there, this is something I've done many times - sip.conf has the correct nat= localnet= externip= settings, the router has ports 5060-5069 and 1-2 forwarded to the internal IP address of the * box. (and 4569 for IAX, but we're just using SIP here) The * server has a few internal (LAN) and external SIP phones, but also has 2 SIP connections to an external PSTN provider. I don't know what this is as I don't have any control or access to it, but both go to the same IP address with different account details (username/passwords) Both these SIP - external PSTN provider connections register OK on the * box, and outgoing calls placed over either connection works perfectly. Outgoing callerId (set by the external provider) works as expected. ) I have dialling prefixes for each 'line', nothing special there, that side of it all works as expected. The problem is that only the last one in the sip.conf file actually accepts incoming calls when dialled from the PSTN side. (They have different PSTN phone numbers) If I swap their entries over in the sip.conf file, then the other one takes the calls. When dialling the first number, nothing seems to get through to the * box at all - nothing on the console in verbose mode, nothing in the log-file. The 2 SIP account setups are otherwise identical (generated by a web interface), just the usenrname password differing, and the account name. Anyone seen this before? I'm wondering if it's an issue with the rotuter (Draytek 2800 ADSL), or is there an issue with 2 SIP channels to the same external IP address (port clash?) I've tried with without bindport= settings in the sip.conf file too - doesn't make any difference. Any clues appreciated! Gordon ___ --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 do sip debug and then look again if still nothing then from linux do tcpdump -Avvv host ip-address of problem device and see if its getting blocked by iptables or not even reaching you. You should prolly show us what your sip.conf looks like and the dial command in use as well as the context it is in. ___ --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] Meetme problems
Hi I have reading the voiip side i found some document says The conference bridge runs Ulaw codec by default. If you let people connect with GSM or other codecs, Asterisk will use CPU power to convert audio between codecs iam using vicidial and meetme for callcenter application. iam geting choppy voice, and voice breaks. iam using connecting VoIP SIP provider using g729 codec, since i can save bandwidth iam using client side also g729, so no translation required but after i see this document, will meetme convert the g729 to GSM or ULAW internall, and i have will have cpu load, is this correct. if i dont want to CPU loadup more, i should use GSM or ULAW at client side is this correct. can some one correct me if iam wrong suggestions welcome ram ___ --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] how can qualify=yes trigger some external event?
Watkins, Bradley wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Friday, June 01, 2007 6:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] how can qualify=yes trigger some external event? Hi all, The option qualify=yes allows Asterisk to check if it can reach the peer. If the device does not answer within the time-out period, Asterisk considers the device off-line for future calls. Is it possible to use this feature to trigger some external event, in case of failed reply from the peer that is tried to be reached? How can that be done? Regards, Ricardo. Hi, Ricardo. Currently there is no way to do this in a pure configuration-only sort of way. However, if you're even moderately adept at C a cursory glance through chan_sip.c will show that it would be quite straightforward to modify the code in order to allow this. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 qualify=yes generates events that can be viewed from AMI, they are: 'Event: PeerStatus' 'PeerStatus: Lagged' 'Event: PeerStatus' 'PeerStatus: Reachable' The other fields give the peer name and like, for more details view the chan_sip.c source, the calls you are interested in there are to a function called manager_event(). I personally have an perl script that camps the AMI and alerts me when these events occur. ___ --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] False ring problem
I agree with Eric. The situation gets worse when you comes to know that some bad carriers uses the -r statement to lead the user to think that its call is already ringing when it is, in fact, still looking for a circuit/network to connect Well, in any of those cases, the solution is simple: Stay far from them! Rgds, Ricardo Martins. Eric ManxPower Wieling escreveu: In my opinion, any carrier that adds r to a Dial line without a VERY, VERY good reason is not a carrier that I want to use. Using r is a classic newbie problem. It indicates a serious lack of understanding about Asterisk. Rizwan Hisham wrote: Well, you r right. This was the carrier`s fault. Its been removed on our request and now we r okay. thanx to all. On 5/31/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Rizwan Hisham wrote: Nobody is using r option anywhere in my dialplan, thats 4 sure. And im also not using any PSTN line to connect to outside world. my system is based on voip only. SIP PHONE--ASTERISK--CARRIER-OUT My only idea is that the carrier might be using the r option. If they are then you should switch carriers. Also callprogress=yes might cause the problem you are experiencing, but I doubt this is the case. ___ --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] multiple host= in sip.conf
David Boyd wrote: On Wed, 2007-05-30 at 15:29 -0500, Eric ManxPower Wieling wrote: Bryan Laird wrote: for inbound connections how does asterisk manage host=host-name returning multiple A records... will it allow authentication for any of the IP's returned? I would assume that in the case of 'inbound' if you specify a host-name that you have PTR records for you could do it in one entry again I'm making a blind assumption. As I understand it, Asterisk does a DNS lookup on load/reload and uses whatever the first IP address returned. allow= and deny= is what should be used for access control. Not the host= line. The host= line is normally used for Asterisk - Device stuff. ___ --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 Does that mean that even when dynamic dns entries exist and the time to live is set to 15 minutes asterisk will continue to try using the old expired results? Dave ___ --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 it does mean that, however it updates at the sip registration timeout, the point in which the device re-registers. So make sure your reg timeout in sip.conf and in the device are below 15 minutes and its not an issue. FYI the default timeout is 3600 seconds, 1 hour. ___ --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] asterisk mysql support
Hi all, I've just realized that my asterisk isn't storing cdr inputs in mysql. cdr_mysql.conf is well configured and I don't know what else should i configure. I'm using Xorcom's packages, cdr status shows: voip*CLI cdr status CDR logging: enabled CDR mode: simple CDR registered backend: csv CDR registered backend: cdr_manager CDR registered backend: cdr-custom it doesn't appear cdr_mysql. Any ideas? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --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] Asteris et winsip
Hi Does anyone tried the Winsip software to test Asterisk? _ Appelez vos amis de PC à PC -- C'EST GRATUIT http://get.live.com/messenger/overview___ --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] how can qualify=yes trigger some external event?
qualify=yes generates events that can be viewed from AMI, they are: 'Event: PeerStatus' 'PeerStatus: Lagged' 'Event: PeerStatus' 'PeerStatus: Reachable' The other fields give the peer name and like, for more details view the chan_sip.c source, the calls you are interested in there are to a function called manager_event(). And so you are right, I wasn't thinking about AMI for some reason. Yes, that's an entirely plausible way to have actions performed when the event occurs. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 mysql support
Issue: module load cdr_addon_mysql On the asterisk command line and post any error messages you receive Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Quintana Cruz Sent: Friday, June 01, 2007 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk mysql support Hi all, I've just realized that my asterisk isn't storing cdr inputs in mysql. cdr_mysql.conf is well configured and I don't know what else should i configure. I'm using Xorcom's packages, cdr status shows: voip*CLI cdr status CDR logging: enabled CDR mode: simple CDR registered backend: csv CDR registered backend: cdr_manager CDR registered backend: cdr-custom it doesn't appear cdr_mysql. Any ideas? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --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] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks
John Hughes wrote: For me all these numbers look too small to be useful for benchmarking. John, They are small, and they are probably more useful as baseline numbers. I'm working on writing up some data I've collected off of our production switch. The call range is 0-450 at 10 call increments. Unfortunately, it's a live environment so it's less than ideal for benchmarking. The makeup of the calls varies, I'm relying on historical data (ie. I can't reproduce the scenarios), and my sample sizes are much bigger for 0-300 calls than they are for 300-450. Nonetheless, there is some knowledge to be gained by studying the numbers and I'm sure that 300 calls constitutes large scale for most people. In the future, I'd like to recreate these numbers using something like SIPp to give me more control. Until then, I'm working with what I have. Thank you for your replies, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] SIP NAT ...
On Fri, 1 Jun 2007, Anthony Francis wrote: do sip debug and then look again if still nothing then from linux do tcpdump -Avvv host ip-address of problem device and see if its getting blocked by iptables or not even reaching you. You should prolly show us what your sip.conf looks like and the dial command in use as well as the context it is in. I'll see if sip debug shows anything, but when I did a quick tcpdump earlier I didn't see anything. (there are no iptables, just a router with port-forwarding to the box) It's always the 2nd entry in sip.conf that works - The first one never works. I can swap them round, sip reload and the other one will then work, but never the first one! If I just have one, (either one), it works perfectly well. There are register statements for both accounts, and sip show peers indicates that both are registerd OK. Not sure how the dial command will help you as it's incoming from a foreign system that doesn't work. As far as I can tell, the SIP commands doesn't even make it as far as the box. There is nothing in console output, and callers get a number unobtinable signal. Outgoing dialling is perfectly fine and does what I expect it to do over both lines. I just want to make sure there's nothing amis at my end before I go chasing the external provider. My suspicion is that there is an issue with 2 SIP channels to the same external provider from the same internal IP address - either something to do with NAT handling at my end (useless Draytek router?), or the remote end just not expecting 2 channels from the same IP address (although that would be the scenario with multiple phones inside a NAT fiewwall, but each with their own internal IP address using STUN rather than one IP address opening 2 SIP channels) Doing a full DMZ redirect isn't an option here as there are other servers behind the firewall handling email. Setting up SIP channels this way is something I've done many times before (it's automated on my systems via a web interface, so it hopefully doesn't make typos :) and it works to many different systems, but I've never had 2 going to the same IP address before. I'll do more tests over the weekend though (when the client isn't using their system!) and extract the config files. Thanks, Gordon ___ --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] SIP NAT ...
On Jun 1, 2007, at 9:45 AM, Gordon Henderson wrote: [snip] Both these SIP - external PSTN provider connections register OK on the * box, and outgoing calls placed over either connection works perfectly. Outgoing callerId (set by the external provider) works as expected. ) I have dialling prefixes for each 'line', nothing special there, that side of it all works as expected. The problem is that only the last one in the sip.conf file actually accepts incoming calls when dialled from the PSTN side. (They have different PSTN phone numbers) If I swap their entries over in the sip.conf file, then the other one takes the calls. [snip] I may be mistaken here, but don't you need to use different ports for each line? ie: Port 5060 for line 1 and 5061 for line 2? Tom ___ --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] chan_iax2.so issues
Hi folks We've a few problems with a rebuild of one of our asterisk boxes, same kernel and configs as previously but unfortunately strange iax issues. If we load chan_iax2 then the system hits 100% CPU, if we do not load this module then all is well. I have tried removing the iax.conf and loading the chan_iax2 within the console and I got an error that included: iax2 show cache' already registered (or something close enough) This implies that another module is stepping on chan_iax2's toes. I've checked the loaded modules and none of them mention iax ... Has anyone else come across this issue or can shed some light on the module crossover. For reference the issue happens with both kernels I have tried 2.6.20.4 and 2.6.21.3 and both asterisk 1.4.2 and 1.4.4. Any help appreciated. Regards Simon Alman ___ --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] Audio going blank for a few seconds andthencomes back. What could be the reason?
If its all local network, then I would agree with you. In our situation, we had people using both SIP and IAX over a home high-speed and we ran into the problem I mentioned. We also tried to setup a IAX trunk between 2 locations where one end was on a normal high-speed connection. We would see no more than 2-3 seconds of silence though. Any more than that, and I agree, something much larger is the problem. However, in our case, the connection was the problem. When we did packet trapping, we could see the handful of packets missing, which made sense to us. Since this certainly not this situation though, I would do the packet capturing like everyone else is recommending. Something has to be odd there. Zeeshan Zakaria wrote: Rob, as I mentioned before, here the main trunk is a T1 PRI on which this customer face this problem. Local phones are connected to the Asterisk server on their local network, and then calls go through the PRI. There is a VoIP trunk too only for long distance, and same problem happens there. So I was thinking its the network issue. On 6/1/07, *Rob Schall* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We have the same problem with our system. Unless you have a solid (not just high speed) connection between the 2 parties, you're going to get silence a few times during the call. We had set up a user on a business comcast high-speed, thinking that would be more than enough. Turned out though, with most high speed solutions, there is some limited packet loss and its just to be expected. You internet browsers, etc, would normally just re-request the packet and move on, but with a stream, you're out of luck. The only real solution is to have a dedicated T1 or mpls connection or something like that for perfect quality. We have solid connections between our offices and haven't had a problem yet. Steve Hanselman wrote: You can use tcpdump or ethereal (wireshark now) to capture the stream and then see if there was loss during the call, just leave a capture going then get your users to mark out the time at which they encountered the silence, compare this to the server time (e.g. their watch to the server) to get a time difference, then figure out what time you need to look at in the trace. Steve *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Zeeshan Zakaria *Sent:* 01 June 2007 13:02 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason? There are some remote extensions connected on this system, and calling long distance is purely on voip. These remote extensions also face the same thing, i.e. audio going blank for a few seconds, when dialing long distance. So in this case, no PRI is involved. Its either the server, or the network. Now I don't know how to find out what is it and why? On 6/1/07, *Steve Hanselman* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I think this is more related to the PRI, we've been seeing this for a few weeks now, and our environment is bridged PRI-PRI on the same board. Steve The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendataco.uk http://www.brendataco.uk ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Cisco 7961G
All, I am having a lot of trouble with the Cisco 7961G phones. I have managed to get them up and running with Asterisk to the point where I can get incoming calls and make outgoing calls. The problem is when I make outgoing calls or extension to extension calls, the calls die after 20 seconds. I have google'd around and came up with little that is of help. The firmware version I am using on the phone is 8.0.4SR1. I have tried tcpdumping the conversation and I see that the phone doesn't send the SIP/SDP ACK packet back to the remote end. Sometimes it does, but that's a rarity. There doesn't seem to be any rhyme or reason as to when it will send the SIP/SDP ACK. All I see is the following before the phone hangs up at 20 seconds (201 is the phone and 205 is the Asterisk Box): 10.230103 192.168.0.205 - 192.168.0.201 SIP/SDP Status: 200 OK, with session description Is there a newer version of the firmware that fixes this? Is there a setting in Asterisk that can fix this? Any help is greatly appreciated. Thanks. Eric ___ --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] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
John Hughes wrote: OpenSSI can't (at the moment) migrate threads between compute nodes. It can migrate separate processes, but doesn't Asterisk use threads? John, Asterisk uses 1 thread per call, plus about 10 to 15 background threads that persist throughout the life of the process. I'm curious if the 1 thread per call model is efficient as the number of calls increases. It's possible that in the 100+ call range that there is a significant overhead to managing all of those threads without much gain since most servers have 1 to 8 processors to actually schedule them on. Acquiring locks on shared resources between the threads could be pretty nasty at that point, too. I wonder if pooling the calls in X threads, where X is a value that is determined at compile time by looking at the number of processors available, would be more efficient? This is probably just an academic question, because I'd imagine it would require an overhaul of the codebase to accomplish. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes
I previously worked for a company that did some heavy load testing with Asterisk on multiple core Sun systems. We saw that no matter how many cores you threw at Asterisk, it always used ONE core to process calls, even at very high loads. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew J. Roth Sent: Friday, June 01, 2007 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes John Hughes wrote: OpenSSI can't (at the moment) migrate threads between compute nodes. It can migrate separate processes, but doesn't Asterisk use threads? John, Asterisk uses 1 thread per call, plus about 10 to 15 background threads that persist throughout the life of the process. I'm curious if the 1 thread per call model is efficient as the number of calls increases. It's possible that in the 100+ call range that there is a significant overhead to managing all of those threads without much gain since most servers have 1 to 8 processors to actually schedule them on. Acquiring locks on shared resources between the threads could be pretty nasty at that point, too. I wonder if pooling the calls in X threads, where X is a value that is determined at compile time by looking at the number of processors available, would be more efficient? This is probably just an academic question, because I'd imagine it would require an overhaul of the codebase to accomplish. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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 mysql support
On Fri, Jun 01, 2007 at 10:37:07AM -0500, Diego Quintana Cruz wrote: Hi all, I've just realized that my asterisk isn't storing cdr inputs in mysql. cdr_mysql.conf is well configured and I don't know what else should i configure. The module was indeed not there. Building it. Thanks for the note, Diego. (We were focusing a bit more on sqlite recently) -- 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
RE: [asterisk-users] how can qualify=yes trigger some external event?
Thanks to all, I guess I'll try to use the AMI with some perl script I'll write to trigger an external event. Other option may be using siksak or sipp with some perl script. Wich option should be best or more straitforward? Thanks, Ricardo. Quoting Watkins, Bradley [EMAIL PROTECTED]: qualify=yes generates events that can be viewed from AMI, they are: 'Event: PeerStatus' 'PeerStatus: Lagged' 'Event: PeerStatus' 'PeerStatus: Reachable' The other fields give the peer name and like, for more details view the chan_sip.c source, the calls you are interested in there are to a function called manager_event(). And so you are right, I wasn't thinking about AMI for some reason. Yes, that's an entirely plausible way to have actions performed when the event occurs. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
Hi Matthew:
Your environment sounds quite challenging and I'd be interested in the
analysis of what is limiting the throughput.
I agree that there's no easy way to distribute and single queue across
multiple boxes.
But here is a scaling idea for you. We've used it successfully to
handle a large inbound call centre. It also provides resilience:
1) Incoming PRIs connect to multiple boxes that we'll call the voice gateways.
Each box can have a proportion of your PRIs connected. Depending on
the box power, up to 8 or so.
2) Agent registrations are spread across these same boxes.
3) Lastly you define two or more additional boxes as your queue servers.
Every queue server has defined on it all the queues you need. But for
each queue one server is regarded as the primary and the other as
secondary. You mix things up so in the normal event about half your
queueing calls are on each server (extend the idea for more than 2
queue servers).
Incoming calls on the voice gateways are sent to the Queue server over IAX:
exten = 1234,1,Dial(IAX2/primary1234/${EXTEN})
exten = 1234,n,Dial(IAX2/sec1234/${EXTEN}) ; if we can't get to the primary
Now when an Agent wants to login, you have their agent gateway log in
to both of the queue servers on their behalf, using an IAX2/.. channel
to get back to the agent's voice gateway.
So on the queue server we have the agents for the queue logged in as
say IAX2/voicegw1/6001, IAX2/voicegw2/6002 etc etc.
The trick is to use transfer=yes aka notransfer=no on the various
boxes. So as soon as the call gets connected to an agent it
disappears off the queue box completely.
The nett result is that the queue servers only have to handle
customers who are still in the queue. As soon as they get connected
to an agent the call is directly from the arriving voice gateway to
the agent's voice gateway and on to the agent. In a proportion of the
time that even turns out to be the same box.
You can scale up the number of voice gateways as required and handle
1000s of calls connected to agents without needing supercomputers.
You still handle all the people queueing on a particular queue all on
the same queueing server. So you can tell them where they are in the
queue and all that. But you can split up your queues across multiple
boxes to help divide and conquer the load.
If you can reach the agent phone directly using IAX (use an IAX
softphone or something) you can make a little optimisation and log
IAX2/agentipaddress into the queue directly. Then the call gets
optimised to go directly from the incoming voice gateway to the
agent's PC.
Resilience? If a queue server is down, new callers will automatically
start to queue on the backup box for the queues affected. The agents
are known on both primary and backup queue boxes so things keep going.
If a voice gateway goes down you lose just some of your PRIs, so you
are still in business. If you need the capacity, use an ISDNguard to
kick the PRIs onto one of the other voice gateways. Agents that were
on the voice gateway that went down will need to reregister to a box
still running. IP address takeover can make that happen.
For me this sort of design is much better than one giant box.
Regards,
Steve Davies
Technical Director
Connection Telecom (Pty) Ltd
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Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes
On 01/06/07, Douglas Garstang [EMAIL PROTECTED] wrote: I previously worked for a company that did some heavy load testing with Asterisk on multiple core Sun systems. We saw that no matter how many cores you threw at Asterisk, it always used ONE core to process calls, even at very high loads. This is definitely not true in the general case. But using IAX2 prior to 1.4 does have a limit like that because all network traffic is handled in a single thread. Take a core dump of a working Asterisk box and count all the threads. There's no general lack of multi-threadedness, that's for sure. 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] asterisk mysql support
Speaking of SQLite, is there an Asterisk SQLite command? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, June 01, 2007 9:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk mysql support On Fri, Jun 01, 2007 at 10:37:07AM -0500, Diego Quintana Cruz wrote: Hi all, I've just realized that my asterisk isn't storing cdr inputs in mysql. cdr_mysql.conf is well configured and I don't know what else should i configure. The module was indeed not there. Building it. Thanks for the note, Diego. (We were focusing a bit more on sqlite recently) -- 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 ___ --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] Cutted audio or 2/3s blanks on EuroISDN- Asterisk1.4
On Jun 1, 2007, at 4:20 AM, Steve Hanselman wrote: We're also seeing the same thing, our calls are bridged zaptel calls between ISDN30 PRI interfaces on a single TE410P. We don't' appear to have any lost interrupts. Same as stated, 2-3 second gaps in audio. Make sure that you're using the most current zaptel drivers (1.2 or 1.4). There was a bug introduced a while ago that could have caused audio drops that may have made it into a release. Matthew Fredrickson Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Administrator TOOTAI Sent: 11 May 2007 09:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cutted audio or 2/3s blanks on EuroISDN- Asterisk1.4 Steve Totaro a écrit : Hi Steve Your Zap conf files would be helpful. Zttest results? Cat /proc/interrupts. Sharing interrupts? No. Zap con files should not be relevant as we are using ISDN. [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /etc/zaptel.conf loadzone = us defaultzone=us [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [trunkgroups] [channels] ; context=default ; switchtype=national ; signalling=fxo_ls ; rxwink=300 ; Atlas seems to use long (250ms) winks ; usecallerid=yes ; hidecallerid=no ; callwaiting=yes ; usecallingpres=yes ; callwaitingcallerid=yes ; threewaycalling=yes ; transfer=yes ; canpark=yes ; cancallforward=yes ; callreturn=yes ; echocancel=yes ; echocancelwhenbridged=yes ; rxgain=0.0 txgain=0.0 ; group=1 ; make these both the same. Groups range from 0 to 63. ; callgroup=1 pickupgroup=1 ; immediate=no [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 109917508 0 0 0IO-APIC-edge timer 1: 12365 0 0 0IO-APIC-edge i8042 8: 444560118 0 0 0IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 12: 11367 0 0 0IO-APIC-edge i8042 14:3944731 0 0 0IO-APIC-edge ide0 58: 0 0 0 0 IO-APIC-level uhci_hcd:usb1, uhci_hcd:usb3, ehci_hcd:usb5 66: 0 0 0 0 IO-APIC-level uhci_hcd:usb2, uhci_hcd:usb4 74:4552211 0 0 0 IO-APIC-level libata 90: 18418187 0 0 0 PCI-MSI eth0 98: 27358592 0 0 0 IO-APIC-level HFC-multi 106: 27358571 0 0 0 IO-APIC-level HFC-multi NMI: 14333691827 1273 LOC: 109917988 109917975 109917950 109917910 ERR: 0 MIS: 0 We use ztdummy for Meetme: [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ sudo ./zttest Opened pseudo zap interface, measuring accuracy... 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.951172% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.951172% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% --- Results after 87 passes --- Best: 99.963379 -- Worst: 99.938965 -- Average: 99.952721 lsmod, zttranscode was loaded, I remove it: [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ lsmod Module Size Used by ztdummy10056 0 tcp_diag6400 0 inet_diag 16784 1 tcp_diag mISDN_dsp 201384 1 hfcmulti 79884 1 mISDN_capi107116 1 l3udss146744 1 mISDN_l2 44616 1 mISDN_l1 17560 1 mISDN_core 88224 6 mISDN_dsp,hfcmulti,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1 capi 23616 0 capifs 11152 2 capi kernelcapi 56640 2 mISDN_capi,capi zaptel197608 7 ztdummy
Re: [asterisk-users] applicationmap on features
On Thu, 2007-05-31 at 23:16 -0300, Tomás Laureano Peralta Tormey wrote: Carlos: In your dialplan setup, have you configured the variable DYNAMIC_FEATURES with the list of dynamic features availables? According to features.conf.sample: Note that the DYNAMIC_FEATURES channel variable must be set to use the features defined here. The value of DYNAMIC_FEATURES should be the names of the features to allow the channel to use separated by '#'. For example: Set(DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3) When I need to use dynamic features, I prefer to export the variable DYNAMIC_FEATURES as a global dialplan variable. Yes I have DYNAMIC_FEATURES=automon#testfeature in the global section of my dialplan. All my phones have canreinvite=no and I have the Tt option in the dial macro for all extensions. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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 and CCM 5.x SIP trunk
On Fri, 2007-06-01 at 10:18 +0530, Vamsi Pottangi wrote: Hi Greg, Narrowed the problem ot be that of codec mismatch ;-) Damn CCM, doesn't provide proper debugs. I have another query with CCM and Asterisk integration. In CCM cluster Phones register to 1st CCM and they fallback to 2nd incase the first fails and 3rd CCM incase even 2nd fails. How can asterisk know on which CCM subscriber the phone is registered to? How to make sure that Asterisk tries all avaiable CCMs to check where the phone is registered. Is there any better way to handle this? Thanks, ~Vamsi CCM handles all of that stuff internally. You will see SIP messages from CCMs coming from all of them all the time. It is always safest to put an entry in sip.conf for all of them in the cluster so * can at least receive calls from any of them. As far as placing calls to CCM, CCM will accept it from *, but may use any of the subscribers to route the call to. Those get set in your route list/group priorities under CCM. If you do not set any priorities, CCM will generally use the publisher of the cluster. I have never had any issues as long as all cluster CCMs were in sip.conf. -Greg ___ --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 mysql support
On Fri, Jun 01, 2007 at 10:26:59AM -0700, Douglas Garstang wrote: Speaking of SQLite, is there an Asterisk SQLite command? Trunk has cdr_sqlite, cdr_sqlite3 and res_config_sqlite (huh? still sqlite2? hmmm). But I understand that many people would like to see sqlite3 better used. E.g.: instead of the ancient berekelydb version we use today for the astdb. There is res_sqlite3.c in asterisk_addons, but it is probably broken, and anyway abandoned. It won't build by default in asterisk-addons and I have not tried to include it in my asterisk-addons deb even long before that. -- 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
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
John Hughes wrote: Matthew J. Roth wrote: As far as Asterisk is concerned, at low call volumes the dual-core server outperforms the single-core server at a similar rate. Outperforms in what sense? At low call volumes the cumulative CPU utilization, expressed as a percentage of available processor, is lower on the dual-core server. This is the expected behavior. What I'm proposing (and hope to back up with numbers in the near future) is that as the number of calls rises to the 300-400 range, the cumulative CPU utilization starts to approach the same number on both servers. Unfortunately, I wasn't collecting as much data when the single-core server was in production so some of this is speculation based on my memory of the system's performance. The environment is also different, because we have added agents so the ratio of calls connected vs. calls in queue has changed. Nonetheless, the dual-core server is not performing anywhere near our expectations. Here is something we recently noticed that may explain why the dual-core server is under-performing at high call volumes. The following numbers were collected off both servers while they were in production. Note that while they have similar cumulative idle values, the ratio of system time to user time on the single-core server is roughly 2.3 to 1, but on the dual-core server it is roughly 19.6 to 1. I'm not quite sure what to make of this, but it seems to be very relevant to the problem. Mon Apr 2 12:15:01 EDT 2007 Idle (sar -P ALL 60 14) (60 seconds 14 slices) Linux 2.6.12-1.1376_FC3smp (4core.imminc.com) 04/02/07 12:24:01 CPU %user %nice %system %iowait %idle 12:25:02 all 14.97 0.03 34.25 0.92 49.82 12:25:020 8.83 0.05 33.60 1.28 56.24 12:25:021 17.50 0.02 34.60 0.57 47.32 12:25:022 19.94 0.02 33.52 1.31 45.22 12:25:023 13.62 0.02 35.29 0.52 50.55 Thu May 10 15:30:01 EDT 2007 Idle (sar -P ALL 60 14) (60 seconds 14 slices) Linux 2.6.12-1.1376_FC3smp (8core.imminc.com) 05/10/07 15:38:01 CPU %user %nice %system %iowait %idle 15:39:01 all 2.47 0.01 48.29 0.00 49.23 15:39:010 2.92 0.00 53.17 0.00 43.91 15:39:011 2.98 0.00 48.68 0.02 48.33 15:39:012 2.47 0.02 48.61 0.00 48.91 15:39:013 2.27 0.00 48.35 0.00 49.38 15:39:014 2.38 0.02 47.38 0.00 50.22 15:39:015 2.37 0.02 46.94 0.00 50.67 15:39:016 2.23 0.02 46.63 0.00 51.12 15:39:017 2.17 0.02 46.54 0.00 51.27 I'm working on a follow-up post that will demonstrate this with some benchmarks for a small number of calls in various scenarios on each machine. However, to our surprise as the number of concurrent calls increases, the performance gains begin to flatten out. In fact, it seems that somewhere between 200 and 300 calls, the two servers start to exhibit similar idle times despite one of them having twice as many cores. What do you mean by idle here? Idle percentage as shown in top's or sar's cumulative view. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] how can qualify=yes trigger some external event?
[EMAIL PROTECTED] wrote: Thanks to all, I guess I'll try to use the AMI with some perl script I'll write to trigger an external event. Other option may be using siksak or sipp with some perl script. Wich option should be best or more straitforward? Thanks, Ricardo. Quoting Watkins, Bradley [EMAIL PROTECTED]: qualify=yes generates events that can be viewed from AMI, they are: 'Event: PeerStatus' 'PeerStatus: Lagged' 'Event: PeerStatus' 'PeerStatus: Reachable' The other fields give the peer name and like, for more details view the chan_sip.c source, the calls you are interested in there are to a function called manager_event(). And so you are right, I wasn't thinking about AMI for some reason. Yes, that's an entirely plausible way to have actions performed when the event occurs. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 There are at least 2 perl MCPAN AMI modules you can download, both work verywell but a little differently the two I mention are event driven so should suit your needs very well. The one I often use is POE::Component::Client::Asterisk::Manager ___ --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] reset Polycom phones remotely
No, this is just reboot -- no factory reset. Rob Townley wrote: On 5/30/07, *Mojo with Horan Company, LLC* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: An answer to your original question: if you can get someone _to_ the phones, on the polycom 30x's you would hold the VolDn, VolUp, DND, and Hold buttons for a while to reboot. For anyone with the 50x or 60x, you would hold the VolDn, VolUp, Messages, and Hold buttons. Moj Moj, is this more of a hard reset to factory defaults? Does cutting the power with a power over ethernet switch do what you need? Forum wrote: I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I need to reset or reformat the phones - if so how can I do this remotely? Can anyone think of a reason why these 2 phones cannot contact the boot server when the other 2 can? Steve ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 ___ --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 Time Card
Although they're not free, cepstral voices are an option. They sound
really nice -- http://cepstral.com/ . They range between $7 and $30.
Moj
Nitesh Divecha wrote:
Thanks Shanon and everyones input...
Finally, got the application working as planned with PHPAGI...
Now the only draw back is the voice... I am using text2wav to prompt all
the questions, but the voice is creepy...
Is their any easier way to replace the text2wav voice with proper
recorded female voice?
Please advice...
Cheers,
Nitesh
Shanon Swafford wrote:
I was messing with something similar one day for a trucking company to
track
progress of their drivers.
It is HIGHLY beta, but should get you started:
## extensions.conf ###
exten = s,1,NoOp(FXO Line is Ringing : ${CALLERID(all)})
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,NoOp(${CALLERID(name)})
exten = s,n,GotoIf($[${CALLERID(num)}=9728311600]?agitest|s|1)
exten = s,n,GotoIf($[${CALLERID(num)}=200]?agitest|s|1)
[agitest]
exten = s,1,AGI(test.php)
exten = s,n,Answer
exten = s,n,Background(shanon-welcome) ; Thanks for calling
press
1 for sales, 2 for support, ...
exten = s,n,WaitExten
###test.php###
?php
set_time_limit(6);
require('/var/lib/asterisk/agi-bin/phpagi/phpagi.php');
$agi = new AGI();
$agi-answer();
$cidnum = $agi-request['agi_callerid'];
$cidname = $agi-request['agi_calleridname'];
$agi-text2wav(Hello $cidname);
$agi-text2wav('We are testing so please call our cell phones. ');
$test = 0;
while ( $test 1 ) {
$agi-text2wav(Enter your Order Number);
$load_num = $agi-get_data('beep', 3000, 6);
$tmp = strsplit($load_num);
$mydata = ;
foreach ($tmp as $value) {
$mydata .= $value . ;
}
$agi-text2wav(You entered $mydata. Enter 1 if this is correct);
$test = $agi-get_data('beep', 3000, 1);
$agi-conlog(Customer Entered: $test);
}
/* Add code here to insert $test into a database */
$agi-text2wav('Goodbye');
// $agi-hangup();
function strsplit($str, $l=1) {
do {$ret[]=substr($str,0,$l); $str=substr($str,$l); }
while($str != );
return $ret;
}
?
Regards,
Shanon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh
Divecha
Sent: Thursday, May 24, 2007 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Time Card
Thanks for your reply,
The basic system would work as follows: -
Method 1
===
An employee would call in to the system and a welcome message is
prompted. After that a employee is asked to enter the employee ID and
PIN number and once verified Employee ID, Caller ID, and time of day
is stored into MySQL DB. By end of the day employee will call in again
to logout from the system and same information is stored into the DB.
Method 2
===
This time employee is verified with Caller ID, so the employee ID and
PIN number is skipped and time of day is logged into the DB.
Is it possible?
Thanks,
Nitesh
ram wrote:
On 5/24/07, *Nitesh Divecha* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hello All,
I have been looking for this solution for quite sometimes
Asterisk Time
Card System. I found some discussion from Digium forum but not
quite
helpful.
Hi
what is the mean of time card system ?
is this kind of attendent system ?
kindly give some more details
ram
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Re: [asterisk-users] Cisco 7961G
we are using 7941 with sip v8.2(2)SR3, it working quite well ;-) Eric Lubow wrote: All, I am having a lot of trouble with the Cisco 7961G phones. I have managed to get them up and running with Asterisk to the point where I can get incoming calls and make outgoing calls. The problem is when I make outgoing calls or extension to extension calls, the calls die after 20 seconds. I have google'd around and came up with little that is of help. The firmware version I am using on the phone is 8.0.4SR1. I have tried tcpdumping the conversation and I see that the phone doesn't send the SIP/SDP ACK packet back to the remote end. Sometimes it does, but that's a rarity. There doesn't seem to be any rhyme or reason as to when it will send the SIP/SDP ACK. All I see is the following before the phone hangs up at 20 seconds (201 is the phone and 205 is the Asterisk Box): 10.230103 192.168.0.205 - 192.168.0.201 SIP/SDP Status: 200 OK, with session description Is there a newer version of the firmware that fixes this? Is there a setting in Asterisk that can fix this? Any help is greatly appreciated. Thanks. Eric ___ --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] Audio going blank for a few seconds and then comes back. What could be the reason?
On May 9, 2007, at 7:29 PM, Zeeshan Zakaria wrote: Its a PRI, no VoIP trunks, so no DSL. This happens only in the office, where phones are connected through the same switch on which data flows for the Internet traffic. But this started happening only few weeks ago. Is there any way that I can check if its the switch or the router? You should make sure somebody didn't start using Kazaa to download music or decided to start watching streaming TV in your office :-) Matthew Fredrickson ___ --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] Audio going blank for a few seconds andthencomesback. What could be the reason?
Try On Jun 1, 2007, at 9:24 AM, Steve Hanselman wrote: There seem to be two threads here that mention multi-second loss with the common part being a PRI, certainly for my situation it's purely PRI as the asterisk box sits in between the telco and another PRI enabled PBX and the calls are bridged between the two. There is no network traffic involved in this case. Not sure where to go with mine though, the load average is nice and low, I don't see any missed interrupts and it's only started happening in the last few weeks since an asterisk upgrade. Latest FC6 kernel, latest yum'd asterisk, zaptel etc Not sure whether it's worth pulling a SVN version down and building that, the only issue is I can't currently reproduce this on demand. Be sure to check the latest svn version of zaptel :-) Matthew Fredrickson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: 01 June 2007 14:36 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Audio going blank for a few seconds andthencomesback. What could be the reason? On Friday 01 June 2007 9:24 am, Rob Schall wrote: comcast high-speed, thinking that would be more than enough. Turned out though, with most high speed solutions, there is some limited packet loss and its just to be expected. You internet browsers, etc, would Limited packet loss != **EIGHT SECONDS** of network breakage. Jitter buffers and PLC takes care of most normal network indiscretions, but period dropouts of that big of a time aren't normal and indicate a bigger issue, either with the hardware or the link itself. normally just re-request the packet and move on, but with a stream, you're out of luck. The only real solution is to have a dedicated T1 or mpls connection or something like that for perfect quality. We have solid connections between our offices and haven't had a problem yet. I have numerous installations using standard telco (Bell Canada and Telus) DSL, and at least one on Rogers cable here in Ontario. No real problems. The odd problem if the pipe gets saturated but careful design and monitoring can take care of most of these problems. I agree with Mr. Hanselman; get a packet logger on the link and see what's really going on. Until that's done, everything here is just speculation. I have seen bugs in the IAX2 and SIP jitter buffers on Asterisk which cause dropouts like this, and I'd like to see what's actually going on before pointing any fingers. -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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --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] SIP NAT ...
On Fri, 1 Jun 2007, Tom Rymes wrote: On Jun 1, 2007, at 9:45 AM, Gordon Henderson wrote: [snip] Both these SIP - external PSTN provider connections register OK on the * box, and outgoing calls placed over either connection works perfectly. Outgoing callerId (set by the external provider) works as expected. ) I have dialling prefixes for each 'line', nothing special there, that side of it all works as expected. The problem is that only the last one in the sip.conf file actually accepts incoming calls when dialled from the PSTN side. (They have different PSTN phone numbers) If I swap their entries over in the sip.conf file, then the other one takes the calls. [snip] I may be mistaken here, but don't you need to use different ports for each line? ie: Port 5060 for line 1 and 5061 for line 2? Well, this is something I'm not 100% sure about. Sip.conf has port= and bindport= parameters, and as far as I can make out port= means use this port to connect to the remote server, and bindport= means listen to this point for incoming calls. port= didn't work. Not surprisingly because the remote server is listening on 5060 only. bindport= (and I tried differnet ports for each account - 5062 and 5064) didn't seem to make any difference. Whether this was being absorbed by the NAT functions on the Draytek, I don't yet know. Will do more experiments over the weekend. Thanks, Gordon ___ --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 7961G
On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote: we are using 7941 with sip v8.2(2)SR3, it working quite well ;-) Eric Lubow wrote: All, I am having a lot of trouble with the Cisco 7961G phones. I have managed to get them up and running with Asterisk to the point where I can get incoming calls and make outgoing calls. The problem is when I make outgoing calls or extension to extension calls, the calls die after 20 seconds. I have google'd around and came up with little that is of help. The firmware version I am using on the phone is 8.0.4SR1. I have tried tcpdumping the conversation and I see that the phone doesn't send the SIP/SDP ACK packet back to the remote end. Sometimes it does, but that's a rarity. There doesn't seem to be any rhyme or reason as to when it will send the SIP/SDP ACK. All I see is the following before the phone hangs up at 20 seconds (201 is the phone and 205 is the Asterisk Box): 10.230103 192.168.0.205 - 192.168.0.201 SIP/SDP Status: 200 OK, with session description Is there a newer version of the firmware that fixes this? Is there a setting in Asterisk that can fix this? Any help is greatly appreciated. Thanks. Eric Anything older than 8.0.4SR2 is asking for grief. You cannot even download older from Cisco's website anymore. Those were their CallManager transitional loads from SCCP - SIP that were riddled with bugs. -Greg ___ --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] SugarCRM Integration
Hi folks, I was wondering if there's a guide on how to configure sugarCRM Integration with Asterisk. I was looking in google and all i found was about Trixbox, which has sugarcrm integrated by default. Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --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: OpenVox A400P01on thin client?
On Fri, 1 Jun 2007 14:46:14 +0800, in gmane.comp.telephony.pbx.asterisk.user you wrote: The Openvox A400P01 is not a full length PCI card. It's a half-length PCI card. You may be referring to the Openvox A1200P (12 port) and that is a full length card. Yup, that's what I figured by looking at the picture and comparing with some small PCI cards that I had. I'll order one to check it out. Thanks. ___ --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] Unicall/R2 for Asterisk 1.4 Available for TESTING
Grate job Moy... i will test it on my PBX tomorrow...
Thanks.
On 4/20/07, Moises Silva [EMAIL PROTECTED] wrote:
Thanks a lot for the fix Humberto.
On 4/18/07, Humberto Figuera [EMAIL PROTECTED] wrote:
Hi Moises,
the Asterisk SVN-branch-1.4-r60989 make a change in the
ast_channel_alloc function:
This is a big improvement over the current CDR fixes. It may still
need refinement, but this won't have as many folks bothered.
here the patch for chan_unicall.c ;p
--- chan_unicall.c.orig 2007-04-18 03:32:17.0 -0400
+++ chan_unicall.c 2007-04-18 03:32:26.0 -0400
@@ -2485,7 +2485,7 @@
}
while (x 3);
-if ( ( tmp = ast_channel_alloc(0, state, 0, 0, chan_name) ) ==
NULL)
+if ( ( tmp = ast_channel_alloc(0, state, 0, 0, i-accountcode,
i-exten, i-context, i-amaflags, chan_name) ) == NULL)
{
ast_log(LOG_WARNING, Unable to allocate channel structure\n);
return NULL;
--
Humberto Figuera - Using Linux 2.6.20
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603
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Re: [asterisk-users] ZAP inbound/outbound connection taking too long
On 01/06/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 1 Jun 2007, Gavin Henry wrote: Dear all, I think this is common, or at least how it is supposed to be, but whening dialing over a ZAP channel, it's taking around 5~ seconds to ring on the over end, likewise inbound. This is just with a normal Dial command. It's normal for an analogue Zap channel. Asterisk has to sieze the line (after a basic check to make sure the channel is free), that may entail a delay of a second or so while it makes sure there there is a dial-tone (actually, I'm not sure it waits for a dial-tone), then it sends the digits out via DTMF - that might take a second or 2 for a long number - then it's up to the PSTN switch at the other end to connect the call - depending on the technology, this might take several seconds. What you can do is connect to asterisk (asterisk -r), set verbose , then initiate a dial and you'll see the dialplan progress and you can work out yourself where the longest part of the delay is... Inbound ought to be answered as soon as asterisk hears the ringing signal - but this might be one whole ring time from the ring starting, depending on how caller-id is being handled in your country, again, monitor it by looking at the output on the console, and by connecting an existing analogue phone in paralel with the incoming Zap line. Gordon ___ Thanks for this explaination! Gavin. --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] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
On 01/06/07, Matthew J. Roth [EMAIL PROTECTED] wrote: Mon Apr 2 12:15:01 EDT 2007 Idle (sar -P ALL 60 14) (60 seconds 14 slices) Linux 2.6.12-1.1376_FC3smp (4core.imminc.com) 04/02/07 12:24:01 CPU %user %nice %system %iowait %idle 12:25:02 all 14.97 0.03 34.25 0.92 49.82 12:25:020 8.83 0.05 33.60 1.28 56.24 12:25:021 17.50 0.02 34.60 0.57 47.32 12:25:022 19.94 0.02 33.52 1.31 45.22 12:25:023 13.62 0.02 35.29 0.52 50.55 Thu May 10 15:30:01 EDT 2007 Idle (sar -P ALL 60 14) (60 seconds 14 slices) Linux 2.6.12-1.1376_FC3smp (8core.imminc.com) 05/10/07 15:38:01 CPU %user %nice %system %iowait %idle 15:39:01 all 2.47 0.01 48.29 0.00 49.23 15:39:010 2.92 0.00 53.17 0.00 43.91 15:39:011 2.98 0.00 48.68 0.02 48.33 15:39:012 2.47 0.02 48.61 0.00 48.91 15:39:013 2.27 0.00 48.35 0.00 49.38 15:39:014 2.38 0.02 47.38 0.00 50.22 15:39:015 2.37 0.02 46.94 0.00 50.67 15:39:016 2.23 0.02 46.63 0.00 51.12 15:39:017 2.17 0.02 46.54 0.00 51.27 Have you got, or could you install oprofile? That will give you a LOT of information as to where your CPUs are spending their time, One guess is that you could be hitting contention in the kernel with all the cores contending for some scarce resource. So your cores can't execute because they are waiting on some kernel mutex for access to some resource. That would account for the increase in system time - oprofile would show where in the kernel they are spending time (where those 50%ishes are going). Steve Uhler at Sun has been studying this on his big multi-core Sparc boxes so he can probably contribute some insight. Hope you don't mind a cc, Steve. We're talking about Asterisk/Linux running out of scaling on an 8 core box. 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] Cisco 7961G
It sounds like you are telling me that it is likely a firmware issue and not an Asterisk issue. Would it be possible for someone to provide me with a copy of your SEPMAC.cnf.xml file and whatever other files the phone uses so I can ensure that its not something else? Thanks. Eric On Fri, 2007-06-01 at 15:07 -0500, Greg Oliver wrote: On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote: we are using 7941 with sip v8.2(2)SR3, it working quite well ;-) Eric Lubow wrote: All, I am having a lot of trouble with the Cisco 7961G phones. I have managed to get them up and running with Asterisk to the point where I can get incoming calls and make outgoing calls. The problem is when I make outgoing calls or extension to extension calls, the calls die after 20 seconds. I have google'd around and came up with little that is of help. The firmware version I am using on the phone is 8.0.4SR1. I have tried tcpdumping the conversation and I see that the phone doesn't send the SIP/SDP ACK packet back to the remote end. Sometimes it does, but that's a rarity. There doesn't seem to be any rhyme or reason as to when it will send the SIP/SDP ACK. All I see is the following before the phone hangs up at 20 seconds (201 is the phone and 205 is the Asterisk Box): 10.230103 192.168.0.205 - 192.168.0.201 SIP/SDP Status: 200 OK, with session description Is there a newer version of the firmware that fixes this? Is there a setting in Asterisk that can fix this? Any help is greatly appreciated. Thanks. Eric Anything older than 8.0.4SR2 is asking for grief. You cannot even download older from Cisco's website anymore. Those were their CallManager transitional loads from SCCP - SIP that were riddled with bugs. -Greg ___ --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 -- Eric Lubow LinkExperts, Inc. ___ --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] ZAP inbound/outbound connection taking too long
Gavin Henry wrote: On 01/06/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 1 Jun 2007, Gavin Henry wrote: Dear all, I think this is common, or at least how it is supposed to be, but whening dialing over a ZAP channel, it's taking around 5~ seconds to ring on the over end, likewise inbound. This is just with a normal Dial command. It's normal for an analogue Zap channel. Asterisk has to sieze the line (after a basic check to make sure the channel is free), that may entail a delay of a second or so while it makes sure there there is a dial-tone (actually, I'm not sure it waits for a dial-tone), then it sends the digits out via DTMF - that might take a second or 2 for a long number - then it's up to the PSTN switch at the other end to connect the call - depending on the technology, this might take several seconds. What you can do is connect to asterisk (asterisk -r), set verbose , then initiate a dial and you'll see the dialplan progress and you can work out yourself where the longest part of the delay is... Inbound ought to be answered as soon as asterisk hears the ringing signal - but this might be one whole ring time from the ring starting, depending on how caller-id is being handled in your country, again, monitor it by looking at the output on the console, and by connecting an existing analogue phone in paralel with the incoming Zap line. Gordon ___ Thanks for this explaination! Gavin. You may also have to wait for a dial timeout at the handset while it decides whether you have finished entering digits. Some handsets have a 'dialplan' you can program to recognize common combinations and bypass this delay. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --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] Gateway VIP450FO and VIP 400FO
Hi everyone! I want to know if anyone has the sip gateway VIP-450FO from Planet (www.planet.com.tw). I´m looking for his firmware because I would like to transform my VIP-400FO (H323) in a VIP-450FO (SIP). Does anyone has this firmware to send to me? Thanks, MCelo. ___ --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] OutBound dial plan
Dear members, Would someone please tell me how I can do the following: Let us assume I put out 2 audio files to a directory somewhere. What would be the API call and Dial Plan to pass 3 things: 1) Select the channel to dialout 2) phone number to dial, 3) file path of wav file to play if person picks up, 4) file path of wav file to play if answering machine picks up 5) log the failure/success results of 3 and or 4 Since I can make several API calls like this during the same time, the system will need to queue up the calls as it gets them (unless multiple calls can be made simultaneously). Thanks in advance, -E ___ --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] Call Back Service
Hello Everyone!! Wanted to ask for your help in what is the best way to do a callback service with asterisk. I want to be able to read a file containing two number to call and then call the two numbers and bridge them. Thanks in advance, Costa ___ --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] G729 client and server Side
ram wrote: Hi iam using G729 at server side and same iam using eyebeam with g729 at client side still its take transcoding CPU process or its pass through ram ___ --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 its pass through ___ --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] WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 36458 udptl t38
Hi iam using asterisk1.2.18 in the logs i keep getting this message any help ram Jun 2 05:47:41 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 36458 udptl t38 Jun 2 05:47:41 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 36458 udptl t38 Jun 2 05:54:00 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39218 udptl t38 Jun 2 05:54:00 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39218 udptl t38 Jun 2 05:54:45 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39678 udptl t38 Jun 2 05:54:45 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39678 udptl t38 Jun 2 05:54:54 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39706 udptl t38 Jun 2 05:54:54 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39706 udptl t38 Jun 2 06:01:09 WARNING[890] app_meetme.c: Unable to write frame to channel SIP/1006-b7803008 Jun 2 06:05:21 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 Jun 2 06:05:21 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 Jun 2 06:05:22 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 Jun 2 06:05:22 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 Jun 2 06:05:22 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 Jun 2 06:05:23 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 ___ --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] SugarCRM Integration
I'd like to know as well about this. On 6/1/07, Diego Quintana Cruz [EMAIL PROTECTED] wrote: Hi folks, I was wondering if there's a guide on how to configure sugarCRM Integration with Asterisk. I was looking in google and all i found was about Trixbox, which has sugarcrm integrated by default. Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --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] click to call
So Guys, no go on this topic? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Jueves, 31 de Mayo de 2007 10:58 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] click to call The idea is to put some kind of embedded app on the website so customers with mics can just click an icon or image and connect to our sales people or customer support staff... So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I'm looking for the latter. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Jueves, 31 de Mayo de 2007 10:18 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call Anton Krall wrote: I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? What we have done in the past is created url's like this : sip:4044565941. Xlite will register itself as the sip handler on your system. If you want a generic click to call (ability to call numbers on any given website) check out moziax - Anton Krall.vcf Description: Binary data ___ --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
