[Asterisk-Users] Call Center software opensource or commercial
Hi there, we are looking for an opensource or commercial * based Call Center. Full ACD, call monitoring, multiple queue, IVR, voicemail, management, reporting, CDR, etc is needed. over 100 seat can be the initial target and will grow in a very short time. SIP phones will be used and multiple E1 lines incoming, so to provide full failover a cluster of * machines or some other form of redundancy must be used. I'm sure custom programming will be requiered so offerings are accepted but all work will be done remotely since we are in Central America (unless you happen to live in our country of course...) Any real experiences with * on this? please for commercial offer reply off-list to [EMAIL PROTECTED] since I think the rules of this forum prohibits commercial offerings. So far i have found http://www.aspect.com/ http://www.ebiitech.com/ Thanks in advance, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Center software opensource or commercial
im my case im looking into 100 seats initially and going up to 1000 at the end (over a 18 months period). Looks like we will have to develop *a lot* if we want to use * for it. Maybe a commercial solution will be better at this time. let's see, Cheers. On Tue, 15 Mar 2005 19:16:37 +0100, lenz [EMAIL PROTECTED] wrote: In data Tue, 15 Mar 2005 17:45:18 +0100 (CET), Peter Svensson [EMAIL PROTECTED] ha scritto: Any real experiences with * on this? You can create a quite flexible callcenter solution from ICD (search for app_icd). It is more of a framework to create a call center solution than a finished product. It is increadible flexible though. I know a number of people who built small to medium sized call centers based on * with app_queue, up to nearly 100 seats. Thoug, from what I'm seeing with Xc-Ast clients and prospects, most * based callcenters are in the 15-30 seat range, with 50-70 agents on shifts. Nearly everybody seems to be quite happy with what they've got, at least since * hit 1.0. Cheers, l. -- Assum est, versa et manduca. -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Center software opensource or commercial
And what people are using to deploy super servers with astersik? Itanium with linux? clusters of itanium with linux? or some RISC processor with some *nix? cause it seems asterisk is only 100% supported on Linux/Intel or am i totally wrong? On Wed, 16 Mar 2005 05:51:18 -0600, Rich Adamson [EMAIL PROTECTED] wrote: im my case im looking into 100 seats initially and going up to 1000 at the end (over a 18 months period). Looks like we will have to develop *a lot* if we want to use * for it. Maybe a commercial solution will be better at this time. On Cebit SGI announced a server solution based on Signate software (which is based on Asterisk) that can handle up to 5000 simultaneous calls. I don't know how the marketing drones have cooked up that number but perhaps it's interesting. See http://www.sgi.com/company_info/newsroom/press_releases/2005/march/von.html According to the marketing blurb, The benchmark was a standard SIPP test and was performed by SGI and Signate. The results compared similarly configured systems: an Altix 350 with dual Intel(r) 1.5GHz Itanium 2 processors/400MHz front side bus/2GB memory compared to a dual 3.0GHz Pentium 4 processors/800MHz front side bus/2GB memory. The results based on simultaneous calls terminating with comparable voice quality were 5,002 for the Altix 350 versus 333 for the PC. Its interesting how marketing people leave out the details. The statement only addresses terminating calls (which one is left with the assumption the test only addressed call setup, not teardown, cdr, etc), doesn't mention whether any of those calls could actually carry on a conversation, hints that no other application (eg, voicemail) was in use simultanously, and most likely assumes the equivalent of canreinvite=yes on a local lan segment following call setup. However, the stats do seem to support what many of us have already experienced, and that is the pci bus limitations with some Intel chipsets is far less then reasonable for realtime apps (such as *). It would be very interesting to see some real life stats with a reasonable mix of * apps including voicemail, transcoding, T1s, etc. If the box could actually sustain 5,000 real life simultanous calls, it could replace a hugh percentage of the US class-5 Central Offices (not to mention PBXs). ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] phones with two ethernet ports
Hi there, what phones are available that have two ethernet ports? I want to do some cabling at a new installation and i heard there are such phones (SIP i guess) out there. That way i dont have to run two cat5 to the user desktop. I think 3COM had one but can't find the web site reference for the two port phone thanks, erick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: phones with two ethernet ports
Update: found the 3Com® 3101 Basic Speaker Phone Provides dual port 10/100 switched Ethernet for one-wire connectivity between the phone and a PC any others not so expensive? does these 3com sip phones work with * ? On Sun, 2 Jan 2005 16:35:12 -0500, Erick Perez [EMAIL PROTECTED] wrote: Hi there, what phones are available that have two ethernet ports? I want to do some cabling at a new installation and i heard there are such phones (SIP i guess) out there. That way i dont have to run two cat5 to the user desktop. I think 3COM had one but can't find the web site reference for the two port phone thanks, erick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Enhancing performance and utility of an Asterisk machine
Hi, some questions/comments about performance/utility of * and * hardware I've been reading this list for a few weeks and I think I have compiled the better feelings of the users. please correct me if I'm wrong, still learning * Will be nice to see something like this in a wiki. After being flamed and corrected I will repost clean data. 1- Transcoding is the process of converting from one codec to another. Example from G.723.1 to G.729. 2- It does not matter if you're doing voicemail or a call. If one of the ends does not speak the same codec, transcoding is involved. 3- Transcoding is very CPU intensive and should be avoided when possible. 4- DSP based cards will improve * performance by offloading work from the CPU. 5- If you configure the SIP phones to use the same codec (G.729) . then no transcoding is involved when they talk to each other. 6- If you're doing VoIP to POTS/T1/E1 you're doing transcoding. 6a-If youre using G.711 as the codec and doing VoIP to POTS/T1/E1 you're NOT doing transcoding? 6b- More than 50 calls VoIP to POTS/T1/E1 will kill an * box due to excesive transcoding?? 6bb- unless using quad machines, plenty of RAM and DSP cards? File Codecs What codec should I use to save my voicemail and IVR prompts? Hardware DTMF generation and cut-through detection are features you must get on a card Integrated DSP Echo Cancellation is a must. any other features that I should go out and buy? * compatible hardware of course. Another off-this-topic question i read that the TDM cards from Digium are having some problems. Im just saying what i read. I have no intention to discuss the problems. But if this is true, then * VoIP players like Nufone with their 80+ * server farms.are using what E1/T1/DS3/etc * compatible hardware? are Digium and VoiceTronix the only * compatible hardware? I googled a lot and found will work with opensource apps but do not explicitly say Asterisk. (Shido can you shed some light on this?) Since I am an end user/Unix sysadmin/VoIP learner I think I will ask the forum to give me the task of doing a wiki or update existing wikis and voip-info.org (when we all agree of course). I bought the * book from signate. It's very good but lack some things...I'm waiting for the next revision (if any). -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prefered server hardware
This question is for my own knowledgei have no experience on this electrical area. why do you want to run -48vdc equipment? what's the advantage of doing that? On Tue, 18 Jan 2005 13:58:59 +0100, Daniel Nyström [EMAIL PROTECTED] wrote: What server hardware would you recommend for an Asterisk system which are really critical? The additional hardware will probably be two digium TE110P cards, and an Adit 600 platform. If it's possible to run on -48VDC, It would be great! Are there any experiences with any HP or FujitsuSiemens systems? Or other complete server systems? Thanks! BR Daniel Nyström ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prefered server hardware
so the switches, etc have two power interfaces? one for the -48dc and the other for the AC plug? or you must specifically buy -48vdc equipment? besides being used by telcosany other advantage? power saving? noise? On Tue, 18 Jan 2005 11:35:55 -0800, Scott Stingel [EMAIL PROTECTED] wrote: -48m volt power is often used in telco central office environments, where the C.O. provides a huge amount of battery-backed up power to the switches and to power the local loops in the event of an AC power failure. Regards Scott Stingel Emerging Voice Technology, Inc. www.evtmedia.com Erick Perez wrote: This question is for my own knowledgei have no experience on this electrical area. why do you want to run -48vdc equipment? what's the advantage of doing that? On Tue, 18 Jan 2005 13:58:59 +0100, Daniel Nyström [EMAIL PROTECTED] wrote: What server hardware would you recommend for an Asterisk system which are really critical? The additional hardware will probably be two digium TE110P cards, and an Adit 600 platform. If it's possible to run on -48VDC, It would be great! Are there any experiences with any HP or FujitsuSiemens systems? Or other complete server systems? Thanks! BR Daniel Nyström ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.4 and more ...
Please, development is important (of course) but updated documentation is important too! is there any special guide followed by * developers about how to document funcionality? I can take care (and will do so happily) of trying to update the documentation at asterskdocs but i must obtain the how it works from somewhere. And I am not a full time programmer so reading code to build documentation is not an option. i've seen the use of parameters or config (in the forums) that are nowhere documented in the wiki or the asterskdocs project. Once again i will be happy to contribute by creating/updating docs. Comments? Thanks in Advance, Erick On Fri, 21 Jan 2005 10:14:56 +, Chris Hills [EMAIL PROTECTED] wrote: Matt Riddell wrote: The easiest way to see that changes is to download one of the packages from the Digium FTP site and read the CHANGELOG. Or carry on reading! ChangeLog: Asterisk 1.0.4 -- general -- fix memory leak evident with extensive use of variables -- update IAXy firmware to version 22 -- enable some special write protection -- enable outbound DTMF -- fix seg fault with incorrect usage of SetVar -- other minor fixes including typos and doc updates -- chan_sip -- fix codecs to not be case sensitive -- Re-use auth credentials -- fix MWI when using type=friend -- fix global NAT option -- chan_agent / chan_local -- fix incorrect use count -- chan_zap -- Allow CID rings to be configured in zapata.conf -- no more patching needed for UK CID -- app_macro -- allow Macros to exit with '*' or '#' like regular extension processing -- app_voicemail -- don't allow '#' as a password -- add option to save voicemail before going to the operator -- fix global operator=yes -- app_read -- return 0 instead of -1 if user enters nothing -- res_agi -- don't exit AGI when file not found to stream -- send script parameter when using FastAGI -- Chris Hills IT Services North East Worcestershire College ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.4 and more ...
Who are these guys? do they maintain de * docs? On Fri, 21 Jan 2005 10:44:29 -0400, [=Jorge Boscan Etura=] [EMAIL PROTECTED] wrote: I second that, we should write to Dag and Thias, they compile so much stuff. On Fri, 21 Jan 2005 13:23:37 +0400, VX Lists [EMAIL PROTECTED] wrote: Will be good, if somebody could provide rpms for every release and also rpm's with static compiled chan_oh323 and Asterisk-oh323 modules ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [Jorge J. Boscán Etura] quando omni flunkus moritati Universidad Fermín Toro http://www.uft.edu.ve Linux 2.6.9 i686 running fc2, lu #137000 cell:584185150239 tel:582512569171 asterisk: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New ip billing solution?? any updates?
Hi people, i've seen the wiki looking for a * billing solution but the links point to websites that have not updated their content (or news) section for over a year. Can anyone recommend a commercial-grade (i mean no mompop cdr system) billing solution that can start small and then scalate as traffic grows and tested/used with Asterisk before? commercial or open source links are ok. thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New ip billing solution?? any updates?
Well, in our country (dont know others) we have different plans for residential users, other plans for commercial users, about 8 international long distance phone services that you can select at dialing time and 3 carriers for domestic long distance.Ohh, and our cellphone providers (TDMA/CDMA/GSM) have different rates (guess this is a rate hell but im sure it got to be worst somewhere else) I am looking for something that lets me plan different providers according to route cost and how to configure * to do so, as well as to handle stuff like today we got 500 minutes at 0.01 but tomorrow at 0.007 so routes must know that today this provider is expensive compared to others but tomorrow it might not. Savinovich, do you have some PDF i can see, or demo? Thanks, On Tue, 25 Jan 2005 16:43:07 -0500, Paul Rodan [EMAIL PROTECTED] wrote: www.bicomsystems.com has a pretty nice billing system built into it, and it's Asterisk based. Not sure if they sell it standalone. We use a mom and pop cdr type of system. We modified cdr_mysql.c to separate national/international and incoming toll free calls into a separate mysql database. Then we use a perl script to read it in, as well as a rate table, do the math and inject the amount the customer owes us into our older billing system which sends out the bills. It can adjust for international calls placed to cell phones or regular city calls, match the international destination, etc. It adjusts for each customer by the account code. I didn't think it was too bad. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: Tuesday, January 25, 2005 3:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] New ip billing solution?? any updates? Hi people, i've seen the wiki looking for a * billing solution but the links point to websites that have not updated their content (or news) section for over a year. Can anyone recommend a commercial-grade (i mean no mompop cdr system) billing solution that can start small and then scalate as traffic grows and tested/used with Asterisk before? commercial or open source links are ok. thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialogic Boards
if it is on Linux hardware with * you'll need to get your hands on the Linux drivers for your dialogic board which are not publicy accesible (or are they?). You must have it recognized by the linux system before doing anything to it by modprobe. On Wed, 26 Jan 2005 10:11:17 -0800, James Ellis [EMAIL PROTECTED] wrote: Hi All, I have checked the supported hardware list of boards that will work with Asterisk. What I am curious about is whether or not I need to have the Dialogic software installed and loaded before I launch Asterisk. Thanks. Jim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialogic Boards
im sorry insmod On Thu, 27 Jan 2005 09:22:11 -0500, Erick Perez [EMAIL PROTECTED] wrote: if it is on Linux hardware with * you'll need to get your hands on the Linux drivers for your dialogic board which are not publicy accesible (or are they?). You must have it recognized by the linux system before doing anything to it by modprobe. On Wed, 26 Jan 2005 10:11:17 -0800, James Ellis [EMAIL PROTECTED] wrote: Hi All, I have checked the supported hardware list of boards that will work with Asterisk. What I am curious about is whether or not I need to have the Dialogic software installed and loaded before I launch Asterisk. Thanks. Jim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Developing an IP Phone
DO you know of companies that will re-brand ip (sip/iax) phones? thanks, On Wed, 02 Feb 2005 00:34:23 +1100, Duane [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: I just thought this link might be interesting to some of you. I know it's m$ware but please hold back the flames. http://msdn.microsoft.com/library/en-us/wcetarget5/html/wce50oriDevelopingVoIPPhone.asp That's fine if you want to develop an SIP phone, but if you want an IAX one, you can take iaxclient and compile it as a DLL. I did that and now I'm using it with Delphi. My phone is almost done :) I'll post it here when it's ready (really soon) What about a kylx(sp?) version for linux? -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
Has anyone on this list have a way to contact ServerWorks? they make the mobos for the G4. I dont have a G4 but i do know HP in the G line uses ServerWorks I have to make a full stop ordering on 2 G4 monsters because of this thread...However one friend is using a sangoma card without problems TE410P/ServerWork motherboard combo not working because of bus problems my less than 1 cent On Mon, 31 Jan 2005 20:42:47 +1100, Eric Bishop [EMAIL PROTECTED] wrote: Did anyone get anywhere with this thread? Any HP G4 series servers working? On Wed, 26 Jan 2005 09:46:31 +1100, Eric Bishop [EMAIL PROTECTED] wrote: Has anyone had any luck with this issue and new Asterisk/Zaptel releases (1.05/1.04)? I am still searching for a solution and waiting for that Eureka! moment.. On Thu, 20 Jan 2005 09:20:09 +0100, Tais M. Hansen [EMAIL PROTECTED] wrote: On Wednesday 19 January 2005 23:15, Eric Bishop wrote: Well guys this is truly bizarre. I managed to get a DL360 G3 to show interrupts with FC2 but not FC3. Exact same config and setup proceedure. Ofcourse neither FC2 or FC3 show interrupts with the DL360 G4. I think TE410P is just a flakey card. Anyone got a DL360 G3 going with a TE410P and FC3? I did manage to get a TE110P running on the DL380 G4. Still can't get the TE410P working in the G4 though. Supports your theory. Sadly we're now being forced to look elsewhere for PRI cards. -- Regards, Tais M. Hansen ComX Networks A/S Tel: +45-70257474 Fax: +45-70257374 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Integration Panasonic PBX
will be nice to have this setting posted. Here in Panama we use lots
of Panasonics and that is a nice one to have
Cualquier cosa nueva me la hacen saber por este posting o a mi correo
eaperezh @ gmail.com
Saludos,
On Tue, 15 Feb 2005 13:38:26 -0300, Sergio Veltri
[EMAIL PROTECTED] wrote:
Maximiliano,
We have implemented that solution succesfully several times.
First:
Does your Panasonic support dtmf inband signaling? without that forget it.
Also you need your setup to look like this:
Outside calls ring into pbx. Pbx co lines are forwarded to a group of
extensions set as voice mail extensions in the pbx programming. Those
extensions are connected to asterisk via an fxo card. That way
asterisk can do ivr and voicemail. You also need to program the pbx
having all phones set to forward all calls (when nobody picks up or is
busy) to that group of extensions so that the call comes back to *
carrying the id of the extension with it.
That's it.
If you need some help let us know. We are in Argentina.
--
Sergio Veltri
www.pointhorizon.com
Tel: +5411-5217-1295
Cell: +54-911-5604-4149
Message: 7
Date: Tue, 15 Feb 2005 11:09:04 -0300
From: Maximiliano J. Goldsmid [EMAIL PROTECTED]
Subject: [Asterisk-Users] Integration Panasonic PBX
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=UTF-8
Hi,
I was woredering if you could help me to put into practice this solution.
The idea: Create a IVR-Voicemail
The scene:
PSTN--/6--PBX/12- Internos
|
/4 ports
|
IVR-Voicemail
The Operation:
1)Where a call enters from the PSTN, the PBX flashes and transfer it
to Asterisk.
2)Asterisk receives the call and you head the in the IVR
3)The caller dials the extension number
4)Asterisk will send the call to the extension number dialed before
4.1) if the extension answers, Asterisk should transfer the call and
free the port, leaning the loop formed between the PSTN and the
extension by the PBX and Asterisk ports are left free.
4.2) If the extension doesn't answer or its busy Asterisk will have to
active the voicemail.
For the time being, the inconvenient I've is in the communication with
the PBX, cause Asterisk after sending the sendtdmf loose any contact
with the status of the call.
I need a way to keep control of the extension of the PBX, if it answer
or not or if its busy, so it can passes control to Asterisk, with
another flash command to active the voicemail menu.
This is are example of a dialplan that doesn't works, cause I send the
call to the extension of the PBX, but I don't keep control of the
status of the call but I can't recover it after, cause if I execute
flash again, the control goes back to Asterisk.
exten = s,1,Answer
exten = s,2,Wait,1
exten = s,3,Background(IVR)
exten = s,4,DigitTimeout,4
exten = s,5,ResponseTimeout,4
exten = t,1,Goto(operadora,s,1)
exten = i,1,Playback(invalid)
exten = _1XX,1,Flash
exten = _1XX,2,background(silence/1)
exten = _1XX,3,SendDTMF(${EXTEN})
exten = _1XX,4,background(silence/1)
exten = _1XX,5,Hangup
Thank you
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Erick Perez
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http://counter.li.org/ (Get counted!!!)
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Re: [Asterisk-Users] ATA's
what about the digium S100i, haven't used but any comments? i know it's only one fxs one lan port does g711 also. no g729. On Tue, 15 Feb 2005 10:29:35 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 15, 2005, at 3:17 AM, Voip Business wrote: hello, my experience 1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE 2.- MTA-V102 3.- Sipura spa 2000 4.- Granstream ATA186 SUXs I can't speak so fondly of the Azatel which I had sitting around after a canceling a VOIP service. Maybe I just need a new firmware rev (but they don't exactly make those available at the Azatel site). Plus, the web interface is excruciatingly limited. I mean, you can't even configure echo cancellation. I think the ATA186-L2 is kind of pointless at this stage. It's old hardware...although Cisco did end up issuing a firmware update last year. Still, there's got to be some reason why Cisco as switched to using a Sipura produce (the PAP2)BTW the ATA186 was designed by some of the Sipura folks as well. My choice is still Sipura-branded equipment. There's no way of knowing how often firmware will be released for the Linksys-branded stuff or what level of support there will be. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax hardphone
that phone is SIP not IAX On Tue, 8 Feb 2005 16:05:53 +0200, Doug Reid - Stormcorp [EMAIL PROTECTED] wrote: Try ACT P104 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: Monday, February 07, 2005 1:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] iax hardphone Hi! Is there such a beast yet available? - Digium IAXy - PA168 chipset: http://www.voip-info.org/tiki-index.php?page=PA168 - farfon (only test devices yet) - several products: http://www.iaxtalk.com/ Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone 101
Fix the missing Contact field for SUBSCRIBE and INFO request Add support for upgrading firmware or modifying configuration via http. Support file path for http url. Add logic to detect and decline duplicate IP during DHCP application stage. Add call time ticking display for callee (BudgeTone 100 only) Support file content authentication checking using AES during firmware upgrade Support for release of IP upon detecting the link is down for more than 15 seconds and re-application for IP address as soon as the link is up again Support attended transfer and Replace header Support Proxy-Require header and its configurable content Support pre-scheduled firmware upgrade checking frequency and add control flag to allow or prohibit auto firmware upgrade. Support configurable PSTN access key string Support 2 different Web login screens (1 for end user and the other for admin). The login interface is shared between 2 different user modes but the edit screen is different. Add port forwarding, DMZ and DHCP server related configuration options to end user configuration screen Fix the loss of registration issue Fix the issue that a HOLD initiated by 1 party can be released by the other when the other party presses HOLD and then releases the HOLD. Fixed the extra @ character in From header when user ID is blank. Fix the issue related to negotiating and using the right MTU when remote end uses a smaller MTU (HT486 only) Fix the PPPoE link state monitoring issue if CHAP is used. Fix the issue where our RTP sequence ID is randomly changed when a 183 response is initially received and then a 200 OK response is received. Fixed layer 2 QoS (VLAN and 802.1p) issue Maintain the credential information for all subsequent REGISTER after the initial registration is successful, as opposed to restart challenge-authenticate cycle for each new REGISTER transaction Fix the reset to factory default which is recently broken Increase the timeout value for PPPoE call establishment. This will better accommodate some Chinese DSL modems' slow response. Also reset IP upon detecting the pppoe link is down for more than 15 seconds. Fix the issue where improperly deleting an un-initialized timer can cause timer malfunction Fix the issue that PPP PAP timer interferes with CHAP negotiation Fix the issue related to processing multiple IP addresses of DNS A record response Fix the issue that PCMU is always included in SDP even if it is never configured on HandyTone products Fix a bug to better handle very long Contact header, e.g., 500+ characters long Fix the ptime negotiation issue where we didn't use the default ptime when the remote end responds with a codec that is different from our first offered codec and which has no ptime in its SDP Fix the issue that after firmware upgrade the device should (but previously does not) reboot automatically. On Fri, 18 Feb 2005 12:01:22 -0500, dean collins [EMAIL PROTECTED] wrote: 1.0.5.22 is available for downloading here http://gs-firmware.gratissip.dk/ I don't know why these are available if Grandstream don't update their webpages to indicate newer versions are available. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Wilson Sent: Friday, February 18, 2005 10:56 AM To: Asterisk-Users@lists.digium.com Subject: Re: [Asterisk-Users] Budgetone 101 1.0.5.16 - the latest version. Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 8:14:41 AM What firmware are you running on your 101? On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: Everytime that I make a call to a Budgetone 101 phone. I always see the following: -- Executing Dial(SIP/1001-bac5, SIP/1000|20|tT) in new stack -- Called 1000 -- Got SIP response 302 Moved Temporarily back from 172.22.5.4 -- SIP/1000-465e is busy I can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this? Josh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk
[Asterisk-Users] mini atx and asterisk (EPIA and the like)
Hi, haven't found anything in google's, i wonder if there is a comparative page of what to expect from running * on motherboards like the EPIA and similar ones. Since i have not used *ever* such kind of mini atx form factor boards, I have no clue about their performance. SIP-SIP communications, voicemail SIP-TDM communications, voicemail how may users (SIP hardphones and analog phones via CPE equipment) Thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with g729 codec
sorry to ask, but what does it mean in passthrough mode ? On Fri, 4 Mar 2005 16:37:06 +, Asterisk guy [EMAIL PROTECTED] wrote: G729 will not work without a licensecan't G729 work in passthrough mode without license? if yes, how to configure it work in passthrough mode? On Fri, 04 Mar 2005 08:50:11 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Fri, 2005-03-04 at 13:29 +0100, [EMAIL PROTECTED] wrote: Hello, I´m trying the g729 codec for testing pourpose. Whe I try to make a SIP call from a phone using g729 codec to another phone using another codec, when the destination phone answer, the call hangs up. this happend in both ways. In the asterisk console I get. Mar 4 13:11:35 NOTICE[24572]: channel.c:1724 ast_set_write_format: Unable to find a path from gsm to g729 What does it mean? Could this occur cause I am using the g729 without licence? If i buy a licence could solve my problem? G729 will not work without a license. The error message above told you that asterisk couldn't find a valid path to convert from gsm audio to g729 audio data. Seems that should have been very obvious from the error. It is well documented had you even decided to search. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk patches - location and use
Hi there, I read from the mailing list that people is using some patches to do special things like fax support (or something related) and other stuff that seem very useful. Like the spandsp patch for * fax located at http://www.soft-switch.org/ Also, www.voip.info.org shows hundreds of good links to addons and apps for * The MOH patch (music on hold) http://perxspace.blogspot.com/2004/11/asterisk-with-moh-patch-on-debian.html However sometimes in those emails people just mention the patch and their functionality by name but not by website (sometimes they do mention the website) Is there an * patch repository i'm not aware of? or do i have to google the web or ask here? also, are these patches for old * versions? does * 1.06 needs them? Many thanks in advance, Erick. -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] telephones to use with asterix
Sirs, i just joined the mailing list and i have a question: What kind of phones can be used with asterix (phones with screen). Basically to see whos calling, display the time,etc...Just like normal phones with display screen do. Thanks, Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk hardware selection question
Given the myriad of telehpone cards available I like to ask this forum for the following combination: Asterix on Linux redhat (9.0 or Fedora) 10 analog extension using conventional phones (lets say Panasonic kx-ts3 analog) 4 analog lines coming from our telco So i will need 3 TDM40B (total 12 FXS and none FXO so i can have 2 extra FXS ports for future) and one TDM04B Quad FXO. Right? and what is the Asterisk support for Digital phones? thanks, Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic VFX/41JCT-LS Linux RH9 Drivers
I read in the archives a post from last year about the Dialogic drivers not being free for use with Linux/Asterisk. So, I have a VFX/41JCT-LS to try with * Suggestions? Purchase digium boards is not an option. We want to test the app before buying any other hardware. thanks, erick, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialogic VFX/41JCT-LS Linux RH9 Drivers
sorry but i did not understand your answer. Erick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, June 18, 2004 4:00 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dialogic VFX/41JCT-LS Linux RH9 Drivers On Fri, 2004-06-18 at 15:39, Erick Perez wrote: I read in the archives a post from last year about the Dialogic drivers not being free for use with Linux/Asterisk. So, I have a VFX/41JCT-LS to try with * Suggestions? Purchase digium boards is not an option. We want to test the app before buying any other hardware. So fleaBay it and use the proceeds for the Digium hardware. You'll come out nearly equal. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialogic VFX/41JCT-LS Linux RH9 Drivers
ok Steven, so i dump dialogic. but my question remains. Are there any free-available linux drivers for the * pbx/dialogic or do i really have to dump my card. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: Friday, June 18, 2004 4:22 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dialogic VFX/41JCT-LS Linux RH9 Drivers sorry but i did not understand your answer. Erick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, June 18, 2004 4:00 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dialogic VFX/41JCT-LS Linux RH9 Drivers On Fri, 2004-06-18 at 15:39, Erick Perez wrote: I read in the archives a post from last year about the Dialogic drivers not being free for use with Linux/Asterisk. So, I have a VFX/41JCT-LS to try with * Suggestions? Purchase digium boards is not an option. We want to test the app before buying any other hardware. So fleaBay it and use the proceeds for the Digium hardware. You'll come out nearly equal. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] two questions
Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware at this moment (but i will in the near future). however i have not been able to find a documentation (not so complicated for a newbie) that help me to setup asterisk in this mode. suggestion/comments/flames welcomed. Thanks, Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] two questions
I see that the 100p is a modem with an Ambient chipset. Why does it sell for 80$ in some places? i can get Ambient pci modem down here for 9 dollars. Any difference? On Tue, 7 Dec 2004 10:58:55 +, Jon Lawrence [EMAIL PROTECTED] wrote: On Tuesday 07 December 2004 04:36, Erick Perez wrote: Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. yes, if that voice modem is a x100p or clone (same chipset). question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware at this moment (but i will in the near future). however i have not been able to find a documentation (not so complicated for a newbie) that help me to setup asterisk in this mode. suggestion/comments/flames welcomed. see www.voip-info.org Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and kphone (sip soft phone for linux) on same machine
Hi, i just installed latest asterisk on fedora rc2 and on the same machine i installed a sip soft phone called kphone. Kphone complains about /dev/dsp being used and can't place/answer calls (/dev/dsp is obviously used by asterisk) . how can share my sound card with these two programs? or can i disable the sound card in asterisk so i can use kphone to place/answer calls? BTW kphone uses my asterisk as the voice server. thanks, erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip phone to sip phone errors
Hi, the following logs are being generated while i test sip-to-sip windows software phones. Dec 7 17:05:16 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time Dec 7 17:05:22 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Dec 7 17:05:26 WARNING[-176354384]: pbx.c:1933 ast_pbx_run: Timeout, but no rule 't' in context 'sip' -- Executing Dial(SIP/erick2-db3b, SIP/erick1) in new stack -- Called erick1 Dec 7 17:05:56 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time Dec 7 17:06:02 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Dec 7 17:06:06 NOTICE[-176354384]: rtp.c:420 ast_rtp_read: RTP: Received packet with bad UDP checksum Dec 7 17:06:06 WARNING[-176354384]: pbx.c:1933 ast_pbx_run: Timeout, but no rule 't' in context 'sip' -- Executing Dial(SIP/erick1-4afc, SIP/erick2) in new stack -- Called erick2 -- SIP/erick2-f752 is ringing -- SIP/erick2-f752 answered SIP/erick1-4afc -- Attempting native bridge of SIP/erick1-4afc and SIP/erick2-f752 Dec 7 17:08:13 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 53352 (Non-critical Response) == Spawn extension (sip, 2000, 1) exited non-zero on 'SIP/erick1-4afc' -- Executing Dial(SIP/erick2-efd9, SIP/erick2) in new stack -- Called erick2 -- SIP/erick2-9cf9 is ringing -- SIP/erick2-9cf9 answered SIP/erick2-efd9 -- Attempting native bridge of SIP/erick2-efd9 and SIP/erick2-9cf9 -- Started music on hold, class 'default', on SIP/erick2-9cf9 -- Stopped music on hold on SIP/erick2-9cf9 == Spawn extension (sip, 2000, 1) exited non-zero on 'SIP/erick2-efd9' -- Executing Dial(SIP/erick2-2685, SIP/erick2) in new stack -- Called erick2 -- SIP/erick2-eeae is ringing they always drop the call. suggestions? thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help with mfcr2 and pri
Hi, While I wait for my unresponsive telco to provide some assistance, can you provide some configuration details for the following config? Sangoma 102 (dual E1) card Location: Panama, Central America Telco: Cable Wireless Panama Lastest stable asterisk 1.2.x compiled from sources Site A in one office Site B is another office in another town When I asked the telco about using CAS or CCS and CRC4 or NCRC4 the technician said: what? im not sure what you mean. Normally it should be CAS/NCRC4 with an E1 MFCR2 right? and CCS/NCRC4 with Euro ISDN PRI on E1 right? What stream are you going to use (structured/unstructured) structured G 703; TS 16: Signalling Line core (HDB3/AMI) HDB-3 Leased line length (wireline of G703 trunk) G.SDHSL Channel level protocol(Site a) MFC-R2 Channel level protocol(Site b) Euro ISDN PRI How should I configure my sangoma with this settings? zaptel and zapata? what of the many unicall downloadables should I use? any other questions I should ask to my telco? Thanks, -- Erick. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with mfcr2 and pri
I have received the follwing info from my telco. E1, PRI, CAS, HDB3, dss1 any help? On 7/25/07, Erick Perez [EMAIL PROTECTED] wrote: Hi, While I wait for my unresponsive telco to provide some assistance, can you provide some configuration details for the following config? Sangoma 102 (dual E1) card Location: Panama, Central America Telco: Cable Wireless Panama Lastest stable asterisk 1.2.x compiled from sources Site A in one office Site B is another office in another town When I asked the telco about using CAS or CCS and CRC4 or NCRC4 the technician said: what? im not sure what you mean. Normally it should be CAS/NCRC4 with an E1 MFCR2 right? and CCS/NCRC4 with Euro ISDN PRI on E1 right? What stream are you going to use (structured/unstructured) structured G 703; TS 16: Signalling Line core (HDB3/AMI) HDB-3 Leased line length (wireline of G703 trunk) G.SDHSL Channel level protocol(Site a) MFC-R2 Channel level protocol(Site b) Euro ISDN PRI How should I configure my sangoma with this settings? zaptel and zapata? what of the many unicall downloadables should I use? any other questions I should ask to my telco? Thanks, -- Erick. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming call detected
Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 My /etc/asterisk/zapata.conf is: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no #include zapata-auto.conf Zapata-auto.conf has: callerid=asreceived ;Sangoma A102 port 1 [slot:1 bus:4 span: 1] switchtype=euroisdn context=from-pstn group=0 signalling=pri_cpe channel = 1-15,17-31 Note: According to the tech support in the local telco, my E1 should be: E1 PRI, CAS, HDB3, NCRC4, DSS1 However if I configure the card for CAS, it will never connect. My card is currently configured (and makes only outgoing calls) as: E1 PRI, CCS, HDB3,NCRC4 (i have no idea what dss1 is or where it goes) My /etc/wanpipe/wanpipe1.conf is: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 4 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES thanks for your help. -- Erick Perez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
Yes I do. I even did a pri debug span 1 and when I call the asterisk box, it sees nothing. On 7/26/07, Idris AVCI [EMAIL PROTECTED] wrote: Do you have any extension in default context of your extensions.conf file to accept incoming calls ? It must be something like; exten = 12345678,1,Answer() exten = 12345678,2,Playback(Welcome) ... 12345678 = The DID number you are calling to reach E1 Idris -Original Message- From: Erick Perez [mailto:[EMAIL PROTECTED] Sent: Thursday, July 26, 2007 7:03 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 My /etc/asterisk/zapata.conf is: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no #include zapata-auto.conf Zapata-auto.conf has: callerid=asreceived ;Sangoma A102 port 1 [slot:1 bus:4 span: 1] switchtype=euroisdn context=from-pstn group=0 signalling=pri_cpe channel = 1-15,17-31 Note: According to the tech support in the local telco, my E1 should be: E1 PRI, CAS, HDB3, NCRC4, DSS1 However if I configure the card for CAS, it will never connect. My card is currently configured (and makes only outgoing calls) as: E1 PRI, CCS, HDB3,NCRC4 (i have no idea what dss1 is or where it goes) My /etc/wanpipe/wanpipe1.conf is: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 4 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES thanks for your help. -- Erick Perez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
As it turns out the telco was not routing the calls to us, a little misktake they said after 3 days of being with no service. The line was not CAS, it was CCS, no need to compile unicall. Whatever they meant with your card has to be configured with DSS1 will remain in mystery. Maybe someone here can tell me what they mean. The configuration I previously listed is valid for lines in Panama City, Panama. With the telco being Cable Wireless Panama and the asterisk with a sangoma A102. If there's any Cable wireless tech reading this. Guys, your support s*cks big time. Thanks to all for your kind and prompt help. On 7/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: If you do not have any alarms and PRI debug span 1 still gives you nothing then you need to call your telco and say I'm not getting any Q.931 messages on the D-Channel. Stephen Bosch wrote: Erick Perez wrote: Yes I do. I even did a pri debug span 1 and when I call the asterisk box, it sees nothing. Hmn, well, that's telling. Are you using the correct cable? Is the cable plugged into the correct port on the card? The 102 is a two-port. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7940G licensing with asterisk
Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE injectors/switches be used with this phone? Thanks, -- Erick Perez ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can handle the 7940G ? The 7941G does conform to the standard but it only support SCCP (shame on cisco). On 9/27/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Yes, you need to buy a license if you use it with ANY pbx, whether it is Callmangler or Asterisk or whatever. If you buy one used, then you need to pay to re-license it as well. The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you will need a switch that provides Cisco PoE for it to work. Erick Perez wrote: Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE injectors/switches be used with this phone? Thanks, ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Home Test!
Hola Facundo, saludos desde Panama. If you're running asterisk at home or some other asterisk project and you're only concerned about the ATA, well, a HT-286 (entry level, cheap) is a good start. Yes, there are reported issues with the GrandStream equipment but all the others have issues too (ok ok I know, don't start on this one). Since your home installation is not *mission critical* a HT-286 will be good. So far I can tell you that a voice provider in my country uses HT-286 and HT-486 commercially deployed at customer premises and it has been working prefectly. My girlfriend who is at this moment in Belgium has an HT-286 that I sent to her and the ATA register back to Panama with no problems. No echo issues. Maybe due to line conditions in Argentina you need to try different echo cancellers. Cheers, On 1/23/06, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] Home Test!
I haven't worked with sipura. So I can't write about it. If I stick to the reviews, then it is a good/stable product with some minor/strange/rarely-ocurred issues regarding phantom calls. spanish-onno creas que no hablo español, pero sabes que aqui solo puedes postear en ingles no?spanish-off On 1/23/06, Facundo Ameal [EMAIL PROTECTED] wrote: Erick Muchas Gracias por la respuesta. I'm not using any of that projects, it's my own Asterisk installation onto slackware 10. well what can you tell about sipura ones? 2006/1/23, Erick Perez [EMAIL PROTECTED]: Hola Facundo, saludos desde Panama. If you're running asterisk at home or some other asterisk project and you're only concerned about the ATA, well, a HT-286 (entry level, cheap) is a good start. Yes, there are reported issues with the GrandStream equipment but all the others have issues too (ok ok I know, don't start on this one). Since your home installation is not *mission critical* a HT-286 will be good. So far I can tell you that a voice provider in my country uses HT-286 and HT-486 commercially deployed at customer premises and it has been working prefectly. My girlfriend who is at this moment in Belgium has an HT-286 that I sent to her and the ATA register back to Panama with no problems. No echo issues. Maybe due to line conditions in Argentina you need to try different echo cancellers. Cheers, On 1/23/06, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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 -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] ISAC Codec Support
Besides the codecs that * supports. Is there any ISAC implementation for asterisk available? This is to be used mainly with softphones, i haven't seen any hardphones that support this codec. Thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.3 CentOS 4.x RPMS
im insterested in how to do it too. On 1/26/06, Eric Bishop [EMAIL PROTECTED] wrote: Do you have step by step instructions on how you created these RPMs. I would like to create a few of my own but compiled for my own custom kernel and patchea and am not very familiar with RPM packaging On 1/27/06, Andrew McRory [EMAIL PROTECTED] wrote: Available in the usual place. ftp://ftp.linuxsys.com/pub/releases/CentOS-4.0 This release includes minor spec changes, spandsp 0.0.2pre23, a new Sangoma wanpipe RPM for use with the LSE kernel rpm and an AMP installation document. Best Regards, -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. - http://www.linuxsys.com Located in beautiful Tallahassee, Florida Office 850-224-5737 Office 850-575-7213 Mobile 850-294-7567 ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] 5,000 concurrent calls system rollout question
I have different need. In the same issue Vic presents. It's 3000 concurrent calls from PSTN (E1s) to Voip (gsm). And the other way around. 3000 Voip calls (SIP/H323 gsm) to PSTN. no voicemail, but the user may get 5 seconds of help prompts initially. Thanks, On 1/28/06, Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. 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 ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] ISAC Codec Support
Well, skype. but i was tweaking some code. This is more a question for lab usage. On 1/28/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: Erick Perez wrote: Besides the codecs that * supports. Is there any ISAC implementation for asterisk available? This is to be used mainly with softphones, i haven't seen any hardphones that support this codec. Which softphone supports it? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] 5,000 concurrent calls system rollout question
5k+ simultaneous calls (in/out) are becoming normal with the kind of call centers being opened in my country during the past 24 months (Panama, Central America). Take Dell Corp. for example. the call center they have here is about 3k people taking/making calls (internal, to/from US, Europe, Asia). Other Call Centers are in that figure too. For me, this thread seems a good learning point to calculate how to do that with asterisk. Thanks to the people who answered here. On 1/30/06, Kristian Larsson [EMAIL PROTECTED] wrote: On Mon, Jan 30, 2006 at 12:49:15PM +0100, [EMAIL PROTECTED] wrote: Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds That will only do if you talk a lot with your mother in law! ;-) For the rest of the conversation (those with both speaking): 5000 * 64k * 2 = 640M Indeed you are correct, I'll defend myself with stating that I presumed we were talkin full duplex ;) It should in theory work with a 1Gbits Ethernet, but you would be counting on ca 65% utilization. I would normally plan with 30-40 % utilization and you need 2 for redundancy anyway. Though now you're wrong ;) 65% isn't correct. If you're counting both in and out traffic you'll have to assume that the Gigg card is capable of 1Gbps in each direction thus 2Gbps in total and 640M of 2000G is about 30% or just as much as 320M is of 1G. I don't know the average packet size of a voice RTP packet but I guess it's quite small. Being a network guy I've dealt quite a lot with software routers and a normal Linux machine can forward about 500kpps, and this is mere forwarding if you run this via Asterisk you should probably split that by ten. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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 1.4 and Cisco Phones 7940
I have read the wiki and several other internet documents. Can anyone make a comment as to what kind of functionality will you loose if you use Cisco 7940 phones with asterisk 1.4 things like: MWI, call transfer, conference,etc,etc. I have a customer with 6 of those phones that he like to use with the asteirsk PBX. thanks, -- Erick Perez ___ --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] Asterisknow b5 - trouble registering at voip provider
Hi, there. I have asterisknow beta 5 with the following data: Ip 192.168.0.60 mask 255.255.255.0 gw 192.168.0.1 the router (a linksys) has port forwarded the port udp 5060 and from 16384 to 16482 udp-tcp from the internet to the asterisk machine. the only protocol allowed is g729. Which work fine for the ip phones I already have setup in the LAN. My problem is trying to register to a voip provider. in the asterisknow gui I provide: protocol sip register (checked) host sf2.clarocom.net username (my phone number) password (assigned password) While executing sip show claro91 asterisk*CLI sip show peer claro91 asterisk*CLI * Name : claro91 Secret : Set MD5Secret: Not set Context : DID_ Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : 2029191 MaxCallBR: 384 kbps Expire : -1 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: No Overlap dial : No DTMFmode : auto LastMsg : 0 ToHost : sf2.clarocom.net Addr-IP : 200.105.69.132 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 2029191 SIP Options : (none) Codecs : 0x80100 (g729|h263) Codec Order : (g729:20) Auto-Framing: No Status : Unmonitored Useragent: Reg. Contact : asterisk*CLI asterisk*CLI and when i try to call with my lan phones to the outside via the claro91 trunk, I get asterisk*CLI -- Executing [EMAIL PROTECTED]:1] Macro(SIP/6000-0820e870, trunkdial|SIP/claro91/66944780) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(SIP/6000-0820e870, SIP/claro91/66944780) in new stack -- Called claro91/66944780 [May 13 17:37:40] WARNING[5522]: chan_sip.c:11860 handle_response_invite: Received response: Forbidden from 'Erick Perez sip:[EMAIL PROTECTED];tag=as7eabcb2e' -- SIP/claro91-082127d8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Goto(SIP/6000-0820e870, s-CONGESTION|1) in new stack -- Goto (macro-trunkdial,s-CONGESTION,1) -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/6000-0820e870, ) in new stack == Auto fallthrough, channel 'SIP/6000-0820e870' status is 'CONGESTION' asterisk*CLI If I switch from my asterisknow box to the linksys box (that has two rj11 ports) then the registration is fine. I would like some guidance as to how to properly format the registration string for my provider. thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Queuemetrics and Asterisknow
Can I use queuemetrics with asterisknow? I mean, if I modify the dialplan to use queuemetrics (I still don't know if it's possible), will I loose my changes when the time comes to do a conary update of the asterisknow package? thanks, -- Erick Perez ___ --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: Queuemetrics and Asterisknow
I realized that queuemetrics uses Java. Is java available as an rpath package or do I need to get it from sun? Also, will it break asterisknow? Thanks. On 5/21/07, Erick Perez [EMAIL PROTECTED] wrote: Can I use queuemetrics with asterisknow? I mean, if I modify the dialplan to use queuemetrics (I still don't know if it's possible), will I loose my changes when the time comes to do a conary update of the asterisknow package? thanks, -- Erick Perez -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] connecting a FXS-to-sip 4 port device to an avaya system
Hi, this is a signalling question: I have a 4port fxs-to-sip where i connect standard analog phones. I want to connect this device to an avaya PBX and then the device talks to asterisk via SIP. What signalling do i need the avaya to provide? FXO signalling right, like this? avayaanalog_lines_fxo_signal--(FXS side)FXStoSIPdevice(SIP side)--Asterisk thanks, -- Erick Perez ___ --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] connecting a FXS-to-sip 4 port device to an avaya system
Thanks Jerry. Are the avaya station ports a special type ? On 1/18/07, Jerry Jones [EMAIL PROTECTED] wrote: Connect to the avaya line ports, not station ports. On Jan 18, 2007, at 10:46 AM, Erick Perez wrote: Hi, this is a signalling question: I have a 4port fxs-to-sip where i connect standard analog phones. I want to connect this device to an avaya PBX and then the device talks to asterisk via SIP. What signalling do i need the avaya to provide? FXO signalling right, like this? avayaanalog_lines_fxo_signal--(FXS side)FXStoSIPdevice(SIP side)--Asterisk thanks, -- Erick Perez ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] ATCOM AT 468 manuals and firmware anyone?
Hi there, im looking for another place that provides manuals and firmware updates for the ATCOM AT 468 and their configuration with asterisk. the site www.atcom.com.cn has non functional download links. I have several of these units but it came only with one CD, I misplaced it and I cant remember how to factory reset them and what will be the default password in the GUI. thanks for your help. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] VIA EPIA DeadLock Issues
Via EPIA CN1 as well. Di you find any solutions? On 1/10/07, Raymond McKay [EMAIL PROTECTED] wrote: Greetings, I've been having a large number of deadlock issues lately on chan_sip occurring only on VIA EPIA ML6000 boards. I'm curious if anyone else is having similar issues. My Config (have multiple systems all running the same hardware with the same problem) VIA EPIA ML6000 1GB RAM 80GB HDD Various Digium Cards (T1 and TDM cards) Trixbox 1.2.2 (though running stock asterisk code) Asterisk Versions 1.2.12 - 1.2.14 - with and without metermaid patch Problem seems to happen more on systems that use parking lots. The system will run for around 24 hours or so fine, and then mysteriously, without any errors leading up to it, will stop being able to send calls to the chan_sip. System from that point on reports the following in the logs. Dec 13 12:07:04 DEBUG[16415] chan_zap.c: Took Zap/1-1 off hook Dec 13 12:07:04 VERBOSE[16415] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for '0x9896848', 10 retries! Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for '0x9896848', 10 retries! attempting to stop asterisk from the CLI causes the CLI to become unresponsive and a trace shows chan_sip goes into a mutex_wait state. Anybody seen this? Have a fix? Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Via EPIA channel_find_locked: Avoided initial deadlock
In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] ATCOM AT 468 manuals and firmware anyone?
both not available. but thanks. On 1/28/07, Leif Neland [EMAIL PROTECTED] wrote: Erick Perez wrote: Hi there, im looking for another place that provides manuals and firmware updates for the ATCOM AT 468 and their configuration with asterisk. the site www.atcom.com.cn has non functional download links. I suppose you mean the AG 468 If you can find somebody who still uses Internet Explorer, the links works. The download page used to have a link for a page which worked in Firefox, but not anymore. But anyway, here are the links. http://atcom.com.cn/en/down/userguide/EN/AG-468/AG468_User_ManualGuide.rar http://atcom.com.cn/en/down/program/en/ng_series/ag468_060119_vr41rls.zip Leif ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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: Via EPIA channel_find_locked: Avoided initial deadlock
I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Re: Via EPIA channel_find_locked: Avoided initial deadlock
Hmm. Mantis says that in SVN 51223 it was implemented, im running 51363. However I may be wrong. I will apply that patch and let you know. Thanks for the pointer. should I leave asterisk as -march=i586? or 386? On 1/29/07, Steve Davies [EMAIL PROTECTED] wrote: I would be interested to know whether this http://bugs.digium.com/view.php?id=8376 patch makes any difference. The problem is almost certainly not caused by Centos (which is widely used with Asterisk) or EPIA (which I use lots). Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Re: Via EPIA channel_find_locked: Avoided initial deadlock
you got that while doing SIP/ZAP and parking? On 1/29/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 29 Jan 2007, Steve Davies wrote: I failed to notice that it was included in 51363 - I just checked, and that change is indeed already in. Sorry, my mistake. I generally do not change the -march setting, so I am probably using an i386 default. I get segfaults with the VIA C3 and C7 chips (on CN1000 and other EPIA boards) with I leave it as the defaults. I need the -i586 option. -i686 seems the be the default in the makefile. I understand it's to do with the MMX instructions used in some of the codecs... Gordon Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: Hmm. Mantis says that in SVN 51223 it was implemented, im running 51363. However I may be wrong. I will apply that patch and let you know. Thanks for the pointer. should I leave asterisk as -march=i586? or 386? On 1/29/07, Steve Davies [EMAIL PROTECTED] wrote: I would be interested to know whether this http://bugs.digium.com/view.php?id=8376 patch makes any difference. The problem is almost certainly not caused by Centos (which is widely used with Asterisk) or EPIA (which I use lots). Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] detecting avaya busy tone
n asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 Asterisk is connected via tdm400p to an avaya system to reach PSTN. When a pstn phone hangs-up asterisk seems unable to detect the busy tone and i keep hearing like 20 busy tones until the zap channel get closed. I'm using loopstart to connect the fxo to the avaya. Some suggestions for busydetection? Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] detecting avaya busy tone
This is a G3. And I'm not the avaya operator. What do you mean with 2500 set and CPC? On 1/29/07, C F [EMAIL PROTECTED] wrote: What avaya system is this, if the avaya is configured on the ports to use a 2500 set, then it should do CPC and should work as is. On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: n asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 Asterisk is connected via tdm400p to an avaya system to reach PSTN. When a pstn phone hangs-up asterisk seems unable to detect the busy tone and i keep hearing like 20 busy tones until the zap channel get closed. I'm using loopstart to connect the fxo to the avaya. Some suggestions for busydetection? Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
The following strange conditions is happening while I try to dial a
SIP user from another SIp user.
SIP to Zap dialing is fine, as all 4 users can call PSTN.
I'm using Asterisk SVN-branch-1.2-r51359M
Example: extension 3210 calls extension 3213. They are all registered properly:
chrom01*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
3213/3213 192.168.0.112D 5060 Unmonitored
3212/3212 192.168.0.112D 5060 Unmonitored
3211/3211 192.168.0.112D 5060 Unmonitored
3210/3210 192.168.0.112D 5060 Unmonitored
4 sip peers [4 online , 0 offline]
-- Executing Ringing(SIP/3210-084eaa80, ) in new stack
-- Executing AGI(SIP/3210-084eaa80,
agi://127.0.0.1:4577/call_log) in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial(SIP/3210-084eaa80, SIP/3213)|30|to) in new stack
Feb 3 12:42:25 WARNING[10368]: chan_sip.c:1994 create_addr: No such host: 3213)
Feb 3 12:42:25 NOTICE[10368]: app_dial.c:1056 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
**sip.conf***
**
i have 4 extensions, 3210,3211,3212 and 3213. they are all defined in
sip.conf with the following parameters (just change 3212 for the next
extension and so on).
[3212]
username=3212
secret=3212
type=friend
context=default
nat=no
canreinvite=no
[EMAIL PROTECTED]
disallow=all
allow=ulaw
host=dynamic
language=en
dtmfmode=inband
My dial plan is like this:
The AGI is doing nothing more than simple call logging to MySQL
**extensions.conf**
**
exten = _321[0123],1,Ringing
exten = _321[0123],n,AGI(agi://127.0.0.1:4577/call_log)
exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to)
exten = _321[0123],n,Voicemail,u${EXTEN}
exten = _321[0123],n,Hangup
comments?
--
Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
___
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[asterisk-users] asterisk 1.2 branch revision 53132 failed to compile
type codec_zap.c:389: error: dereferencing pointer to incomplete type codec_zap.c:395: error: dereferencing pointer to incomplete type codec_zap.c:396: error: dereferencing pointer to incomplete type codec_zap.c:397: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:399: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_g723toulaw': codec_zap.c:415: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:437: error: dereferencing pointer to incomplete type codec_zap.c:444: error: dereferencing pointer to incomplete type codec_zap.c:444: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:445: error: dereferencing pointer to incomplete type codec_zap.c:446: error: dereferencing pointer to incomplete type codec_zap.c:452: error: dereferencing pointer to incomplete type codec_zap.c:453: error: dereferencing pointer to incomplete type codec_zap.c:454: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:456: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_alawtog729': codec_zap.c:472: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:494: error: dereferencing pointer to incomplete type codec_zap.c:501: error: dereferencing pointer to incomplete type codec_zap.c:501: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:502: error: dereferencing pointer to incomplete type codec_zap.c:503: error: dereferencing pointer to incomplete type codec_zap.c:509: error: dereferencing pointer to incomplete type codec_zap.c:510: error: dereferencing pointer to incomplete type codec_zap.c:511: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:513: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_ulawtog729': codec_zap.c:529: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:551: error: dereferencing pointer to incomplete type codec_zap.c:558: error: dereferencing pointer to incomplete type codec_zap.c:558: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:559: error: dereferencing pointer to incomplete type codec_zap.c:560: error: dereferencing pointer to incomplete type codec_zap.c:566: error: dereferencing pointer to incomplete type codec_zap.c:567: error: dereferencing pointer to incomplete type codec_zap.c:568: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:570: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_g729toalaw': codec_zap.c:586: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:608: error: dereferencing pointer to incomplete type codec_zap.c:615: error: dereferencing pointer to incomplete type codec_zap.c:615: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:616: error: dereferencing pointer to incomplete type codec_zap.c:617: error: dereferencing pointer to incomplete type codec_zap.c:623: error: dereferencing pointer to incomplete type codec_zap.c:624: error: dereferencing pointer to incomplete type codec_zap.c:625: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:627: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_g729toulaw': codec_zap.c:643: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:665: error: dereferencing pointer to incomplete type codec_zap.c:672: error: dereferencing pointer to incomplete type codec_zap.c:672: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:673: error: dereferencing pointer to incomplete type codec_zap.c:674: error: dereferencing pointer to incomplete type codec_zap.c:680: error: dereferencing pointer to incomplete type codec_zap.c:681: error: dereferencing pointer to incomplete type codec_zap.c:682: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:684: error: dereferencing pointer to incomplete type codec_zap.c: In function `find_transcoders': codec_zap.c:849: error: variable `info' has initializer but incomplete type codec_zap.c:849: warning: excess elements in struct initializer codec_zap.c:849: warning: (near initialization for `info') codec_zap.c:849: error: storage size of 'info' isn't known codec_zap.c:854: error: `ZT_TCOP_GETINFO' undeclared (first use in this function) codec_zap.c:859: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:849: warning: unused variable `info' make[1]: *** [codec_zap.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2/codecs' make: *** [subdirs] Error 1 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780
[asterisk-users] Re: asterisk 1.2 branch revision 53132 failed to compile
same with branch revision 53142 On 2/3/07, Erick Perez [EMAIL PROTECTED] wrote: while compiling svn 53132 of asterisk branch 1.2 gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DZAPTEL_OPTIMIZATIONS -DBUSYDETECT_MARTIN -fomit-frame-pointer -fPIC -c -o app_sms.o app_sms.c gcc -shared -Xlinker -x -o app_sms.so app_sms.o make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2/apps' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2/codecs' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DZAPTEL_OPTIMIZATIONS -DBUSYDETECT_MARTIN -fomit-frame-pointer -fPIC -c -o codec_zap.o codec_zap.c codec_zap.c: In function `zap_framein': codec_zap.c:147: error: dereferencing pointer to incomplete type codec_zap.c:149: error: dereferencing pointer to incomplete type codec_zap.c:151: error: dereferencing pointer to incomplete type codec_zap.c:151: error: dereferencing pointer to incomplete type codec_zap.c:156: error: dereferencing pointer to incomplete type codec_zap.c:156: error: dereferencing pointer to incomplete type codec_zap.c:156: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:159: error: dereferencing pointer to incomplete type codec_zap.c:162: error: dereferencing pointer to incomplete type codec_zap.c:162: error: dereferencing pointer to incomplete type codec_zap.c:162: error: dereferencing pointer to incomplete type codec_zap.c:163: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_frameout': codec_zap.c:187: error: dereferencing pointer to incomplete type codec_zap.c:196: error: dereferencing pointer to incomplete type codec_zap.c:197: error: dereferencing pointer to incomplete type codec_zap.c:198: error: dereferencing pointer to incomplete type codec_zap.c:198: error: dereferencing pointer to incomplete type codec_zap.c:199: error: dereferencing pointer to incomplete type codec_zap.c:200: error: dereferencing pointer to incomplete type codec_zap.c:203: error: dereferencing pointer to incomplete type codec_zap.c:206: error: dereferencing pointer to incomplete type codec_zap.c:207: error: dereferencing pointer to incomplete type codec_zap.c:208: error: `ZT_TCOP_TRANSCODE' undeclared (first use in this function) codec_zap.c:208: error: (Each undeclared identifier is reported only once codec_zap.c:208: error: for each function it appears in.) codec_zap.c:209: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c: In function `zap_destroy': codec_zap.c:223: error: `ZT_TCOP_RELEASE' undeclared (first use in this function) codec_zap.c:224: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:227: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_alawtog723': codec_zap.c:244: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:266: error: dereferencing pointer to incomplete type codec_zap.c:273: error: dereferencing pointer to incomplete type codec_zap.c:273: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:274: error: dereferencing pointer to incomplete type codec_zap.c:275: error: dereferencing pointer to incomplete type codec_zap.c:281: error: dereferencing pointer to incomplete type codec_zap.c:282: error: dereferencing pointer to incomplete type codec_zap.c:283: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:285: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_ulawtog723': codec_zap.c:301: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:323: error: dereferencing pointer to incomplete type codec_zap.c:330: error: dereferencing pointer to incomplete type codec_zap.c:330: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:331: error: dereferencing pointer to incomplete type codec_zap.c:332: error: dereferencing pointer to incomplete type codec_zap.c:338: error: dereferencing pointer to incomplete type codec_zap.c:339: error: dereferencing pointer to incomplete type codec_zap.c:340: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:342: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_g723toalaw': codec_zap.c:358: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:380: error: dereferencing pointer to incomplete type codec_zap.c:387: error: dereferencing pointer to incomplete type codec_zap.c:387: error: `ZT_TRANSCODE_MAGIC' undeclared (first
Re: [asterisk-users] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
Indeed. The problem was the ).
thanks to all who helped me debug this...my eyes are not so young anymore...
On 2/3/07, jacobso1 [EMAIL PROTECTED] wrote:
hi,
i think the problem is here :
exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to)
|
replace with
exten = _321[0123],n,Dial(SIP/${EXTEN},30,to)
note, i removed the parenthesis ')' after the {EXTEN}
this should do
regards,
jacobson
---
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http://www.scarlet.be
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[asterisk-users] WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/4-1'
As everybody must be watching the superbowl. I post this to let you have some fun while thinking what this can be. TDM400p (fxo) connected via loopstart to ports in an AvayaG3 call comes in from the avaya to the tdm card: WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/4-1' but call can be processed normally. comments? -- Erick Perez ___ --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 and multiple cpus/cores
I have found a site that list the following (no date in the post, so it may be old): since all transcoding and calls still go through one core in asterisk, it doesn't make sense to buy a multi-core or hyperthreaded system that will only slow you down Does that still applies in asterisk 1.2.14/1.4.x ? Or do we have to tweak source code to balance loads (transcoding,etc) between cores? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Issues with a Linksys SPA 2102 and asterisk
Topology: analog_phone-SPA2102-Navini_Wireless_Router--ISP--Asterisk A ping against the asterisk server shows aprox 145ms roundtrip. 128kbps upstream 512kbps downstream g729a as codec signal quality of the navini router: 100% The ATA operates correctly in every form, however sometimes when someone is talking to me (the other person is at pstn) and then I start talking the other end receives garbled voice and i need to start talking again. So I played with the jitter buffers in the available modes (low, medium, high) (direction upward, downward both) and it seems i cannot improve my voice experience. Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14? thanks, -- Erick Perez ___ --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] Issues with a Linksys SPA 2102 and asterisk
where to change packet size? On 3/9/07, Luki [EMAIL PROTECTED] wrote: Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14? They work fine with Asterisk; most likely it's your wireless link that's the cause of your problem. The jitter buffer will only affect received audio, i.e. on your side, and since that is fine, you probably don't need to adjust it. Instead try this: 1) Change packet size in increments of 20 ms (i.e. 0.02, 0.04 or perhaps 0.06). Your wireless link may not like too many small packets. 2) Turn off silence suppression if it's on. 3) Try a different codec -- g726-32 or even ulaw to see if it makes a difference. See if that helps. --Luki ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] connecting 66 analog phones to asterisk - hardware suggestions
Hi, I am looking to connect 66 analog phones to an asterisk box. I was thinking of a Xorcom astribank 32port (2 of them and another 8 port). this is because the phones have no near connection to an ip network, so replacing the phones in favor of voip phones+network cabling is kinda out of the question. In your experience, will these units support all the phones talking at the same time with other units on the astribank, as well as to the pbx, pstn, etc? The asterisk pbx will be a server-class Hp Proliant unit (potentially a dl320). i must make sure the astribanks will not die when fully utilized. other hardware suggestions for this task will be nice. thanks, -- Erick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk across a firewall
Excuse my ignorance but if i have an asterisk in a LAN, and i have users in their homes/internet (dozens), in order to correctly connect those users across my firewall, what is the technology that i need to buy, called? secure border gateway? session controller? secure gateway? the audiocodes site seems to have many names for the same thing...but i better ask here and learn before i make a big mistake. my customer has a dumb firewall (not SIP aware) that will not replace. he wants another box to do the magic. -- Erick Perez Cel +(507) 6675-5083 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk across a firewall
On Wed, Feb 11, 2009 at 1:56 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Wed, 11 Feb 2009, Erick Perez wrote: Excuse my ignorance but if i have an asterisk in a LAN, and i have users in their homes/internet (dozens), in order to correctly connect those users across my firewall, what is the technology that i need to buy, called? secure border gateway? session controller? secure gateway? the audiocodes site seems to have many names for the same thing...but i better ask here and learn before i make a big mistake. my customer has a dumb firewall (not SIP aware) that will not replace. he wants another box to do the magic. I have many customers like that, and working from home is gaining momenting where I live... So the scenario (if I interpret it correctly): Asterisk at HQ is behind a NAT firewall with remote users (who themselves may be behing a NAT firewall) HQ needs a static IP address on the outside and plenty of bandwidth. The dumb router at HQ needs to port-forward external port 5060 and 1-2 into the asterisk box (you can limit this range - see rtp.conf) Most dumb routers can port-forward. Asterisk needs to know it's LAN and extneral ip address - sip.conf, externip= and localnet= remote extensions need nat=yes in sip.conf and that's basically it. If the remote extensions are themselves behind a NAT firewall, then the easiest way to get them through it is by using a stun server - ether run your own, or use someone elses... Do not do any port-forwarding at the remote users sites. Yes, you can fiddle about with proxies, gateways, etc. but keep it simple to start with and I have many installations doing it this way and it just works. One day I'm sure I'll trip up, but until then... Pitfalls - the same with all VoIP - bandwidth, espeically outgoing b/w from HQ. Broken NAT gateways, and routers which have SIP ALGs built in which are also broken. (Turn them off!) Routers with broken SIP ALG are the biggest PITA to work round. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you all for the excellent responses. I will do some test here to decide on a method/technology to use. -- Erick Perez Cel +(507) 6675-5083 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions
Hi all, thanks for the excellent information about the banks and usb banks. some tech details will prevent us from using usb units. The trunks will be 500 feet away from the new location of the ip-pbx so we have decided to go with channel banks for the trunks and sending the E1 signal over cat 5 (E1 signal can travel un-repeated over 5000 feet) So far we are reading/evaluating about rhino channel banks and a quad E1/T1 (pci-e) on the asterisk box. thanks again -- Erick Perez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 400 calls at g711 how much cpu power
We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? -- Erick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 400 calls at g711 how much cpu power
I totally agree with you Jeff, however some of us do not actually sell viagra over the phone. This is a campaign to spread a message to the population about the health prevention steps that should be taken in order to prevent diseases that are affecting our population. I do understand all of you to be reluctant to help with this post. However judging before listening has been the most devastating problem humans have. We simply do not trust each other. However, just for the sake of posterity: Hardware/Software just one server Dell 2950 / 4GB RAM / four 72Gb ultra320 SCSI hard disks built as RAID-0 Debian as the OS (in 32 bit mode) Asterisk 32 bit 1.4 compiled manually (codecs removed, modules removed,etc, a ton of pure CRAP out!) Only g711/SIP was used 20 second clip was served from ramdisk Dialer: SmoothTorque (those guys simply ROCK!)( setup outbound mode ONLY!) Network: 50 Mbit fiber link to telco provider. Pure IP, no QoS. We were pumping 3k calls-setup/second to the session controller at telco's side. Until we reached controller's max of 10k calls. Server load was NEVER above 3.2 thanks to all for your help. On Thu, Apr 2, 2009 at 7:36 PM, Jon Pounder j...@inline.net wrote: Erick, how about posting your home phone number here so we can all call you and play a 20second audio clip - I am sure you would see nothing wrong with that would you ? ContactTel Business wrote: Your right, i don't think we would help someone asking on advice to send 1 million emails for Viagra would we ? So why the hell aren't we thinking straight and tell the poor guy? Ive seen dialer app that where legit, even worked on some for the military. But this is just spam /pham (phone spam) send 10USD to my email ;) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: April-02-09 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 400 calls at g711 how much cpu power My only comment is that I am having moral issues with assisting anyone that is planning to call one million phone numbers to play a message and hang up. Doesn't sound like an opt-in kind of campaign to me. When such a thing happens to me on my home phone I get extremely angry. j On Wed, 1 Apr 2009, Erick Perez wrote: We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? -- Erick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Cel +(507) 6675-5083 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 400 calls at g711 how much cpu power
I am fairly certain he was simply reporting the results (for posterity) of the event having already happened. Good to know (I guess?) that such small hardware can acheive the performance that was squeezed out of it. Impressive. All THAT said, I am unconvinced that there was no sales effort involved in sending out millions of unsolicited calls. Claim if you like that this was some public information event (which you fail to expand much upon) and convict me of mistrust, but who would have paid for such a thing. TV ads, radio spots, billboards, etc., are much more effective for public information. Unsolicited calls on that order mean only one thing to me - SPAM. So what wonderful product were you informing the public about with regard to the looming threat of illness? Jeff, indeed i was posting for posterity. Maybe someone will benefit in an outbound-only scenario that he/she will not need a supercomputer to pump a 20sec audio clip. Again, this was a public service. And indeed TV and radio was used. Unless you live in a bubble, you may have heard about AH1N1 virus. Which unfortunately hit us (Panama, Republic of Panama, Central America) very hard. I foud very repetitive to tell in my posts that i am from panama, central america, blah,blah blah. Anyways, a quick google search of this forum will also revealed that i am kind of a regular poster and even my cellphone is listed here (Jon Pounder, my cellphone is +507 6675 5083 in case YOU want to sell me a car loan, i dont mind getting a call. Im a IT consultant and i have a chargeback line. Please call me as many times as you want...please do so between 10pm and 6am where my chargeback is the most expensive). Guys, Grow up! Next time someone needs to learn mouth-to-mouth and CPR lessons, please DONT teach him. Because, following your inmature way of thinking, the person who wants to learn CPR may as well be looking for information to learn how to suffocate people. Next time your son wants to know how gasoline works or how is being produced. Please keep your familiy in ignorance. You may be training the next crazy person who will burn things all around the world. But, you wont do that, do you? Again, I always tell my familiy that keeping others in ignorance is bad. but sometimes it must be done for the sake of a greater good, and my comment is always followed with good and sound examples (atomic technology, viruses, etc). But I forgot that Asterisk, the phone lines and a calling system is the way the world is going to be dominated by the martians. So the secret about phone system calculations must be keept in Area 51. Now I understand Kevin Mitnick. Cheers to all. Bye. Erick Perez Cel +(507) 6675-5083 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Bandwidth calculations and PCI/PCIX/PCIE
I found this interesting but old white paper at Dell.com tech solutions and another one from INTEL. It compares bandwidth usage of a PCI, PCI-X, PCI-E in33/66/100/133mhz bus and different technologies that can saturate the bus. It helped me understand the bandwidth required for TDM (sangoma/digium) cards and how far can I push the PCI bus in an old and newmotherboard. I hope it help others to understand how much a network card can pump and make calculations about consumptions in TDM cards. make sure the link is a one-line in your browser Original online document http://www.dell.com/content/topics/global.aspx/vectors/en/2004_pciexpress?c=uscs=08Wl=ens=bsdv here is the link to the same Dell article but in PDF form. http://www.dell.com/downloads/global/vectors/2004_pciexpress.pdf Another interesting document from INTEL www.intel.com/technology/pciexpress/devnet/docs/WhatisPCIExpress.pdf The facts learned from these documents are: a- 3.3volts/32bit PCI cards can be used in PCI-X slots. (i just discovered that, sorry forliving under a rock) b- The slowest PCI card in Mhz will dictate that PCI-X bus speed. So avoid degradation by not installing a PCI card and a PCI-X card in the same bus (check you motherboard design), your motherboard design usually have two buses. c- If you use a PCI-X based implementation motherboard, you will not saturate the bandwidth of the board, using Quad or Octal port cards (e1/t1/j1). -- Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] CPU configuration for 250 calls SIP to SIP to IAX and fonebridge and two asterisk servers
Hi, I would like to read your comments for the following setup: Building A: 3 voice E1incoming toa quad redfone fonebridge (TDMoE) The fonebridge goes to a port in a 24 port gigabit switch in the gigabit switch VLAN1 is for the fonebridge and the first gigabit NIC on a dual NIC server in the gigabit switch VLAN2 is for the second gigabit NIC card on the server andeleven 10/100 switches with 250 SIP phone users running g711 codec (24 phones per 10/100 switch,each switch is 24port) Building A and Building B are connected over a 10Mbits fiber link. Numeric Extensions at building A are 1xxx Building B: same config E1/switch/users as building A Building A and Building B are connected over a 10Mbits fiber link. Numeric Extensions at building B are 2xxx The asterisk servers at each side will talk IAX2 between each other for building-to-building call transfers. Suggested machine: Im considering a Dell PowerEdge 9G 1950, Dual Xeon 3.20Ghz, 1066 FSB, 4GB ram. two 73GB SAS 15k RPMs hard disk and dual gbit network card. Asterisk Features: Music on hold call transfer call waiting (but only on executive phones, around 20) voicemail a small queue (about 10 persons) and a simple IVR (play prompts for department selection, transfer according to selection). No call recording requested at this time. Operating System: Centos 4.3 Codecs: G711 for the SIP to asterisk and IAX for server to server transfers. If IAX is not recommended, please advice. Notes: a- Is is expected to have the 250 SIP users talking either to each other and/or to the other building and/or to the fonebridge E1s. b- I know that for SIP-to-ZAP a calculation of 30Mhz per voice channel is a rule of thumb, but i also read somewhere that the same calculation does not apply when doing Pure IP, no SIP/ZAP and pure g711 implementations I'm in that category. c-Just for the record, what if I change to g729? d- It is expected to have 80% of the calls over the E1 being incoming from the PSTN and the other 20% ar the SIP users calls to the PSTN Is is also expected to haveone 24 port Rhino FXS channel banks connected to the 4th port of the fonebridge. Is used, it will add another 24 users to the setup. Thanks in advance. Your comments are welcomed. Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de PanamaCel Panama. +(507) 6694-4780 ___ --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 much SIP calls can I squeeze from this box
Hi lists, I would like to know how much can i get from the below configuration. I have a machine in my office that I want to use for demo purpose. The features I want to implement are: voicemail (users call the box to get their messages) voicemail to email (some users will the the vm by email) pbx like behavior (music on hold, a simple IVR to select what department to talk to) Full 100% call recording. Software spec: Centos 4.4 Asterisk 1.2.12.1 no sql SIP users with IP hardphones running g711 Hardware: Asterisk Box: Dual core Pentium D at 2.4ghz, 533fsb, Intel 945GNT board,100Mbit intel NIC. Dual 80gbit sata2 disk. A 8-port fxs card (pci in a PCI-X slot) and the FXS will be connected to a Panasonic PBX Protocol: G711 all the way if possible (even in moh) SIP users?: Here it comes my question in terms of: - Registered users - Simultaneous calls (remember full call recording) BTW: What options do I have to minimize disk writes for the call recording part? more ram to make it as a ramdisk? special ramdisk cards? any special format or way to capture/encode/store the recorded stream? During night hours I was thinking of moving the recorded files to another server via NFS. thanks in advance. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Mini call center only 15 seats fxs to sip suggestion
Hi, I looking for an affordable (maybe used) FXS to SIP media gateway (or another method) to be deployed in a mini call center. The final user already has analog phones and a cabling setup in place. The cheap gateway will send and receive SIP traffic to an asterisk box that is already in place and connected to PSTN. The asterisk is there because it will provide voice recording and voicemail to email and a simple IVR. The final user does not want to spend the money associated with items like and audiocodes gateway or a sngoma remora or digium FXS card. that's why we are looking for a media gateway. Since he already have some analog panasonic phones, he does not want to purchase Ip phones. if you have some other ideas, let me know. Ebay turned nothing in my searches. Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Are you using app_meetme or app_conference
Hi, for call centers with voip phones and calls coming in via SIP and Zap, what app_ are people using to do: -conference -listening to conversation of agents Is app_meetme or app_conference? Does app_meetme still suffers from the need to transcode to slin? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 RT on Disk On Module Performance and Durability
Hi, Im doing some research for Disk on a Module (DOM)with asterisk realtime. To have no moving parts for a special project, I know I can use 3.5 or 2.5 HDDs but DOMs sound interesting. Does someone have working experience with this? Basically the Asterisk Realtime will be stored in MySQL and the DB will be stored in a Disk on a Module. I have read that the usual standard is 2,000,000 MTBF and 2,000,000 Read/Write Cycles. Is there an utility/section/procedure that can count/display the reads and writes a normal Linux system does? That result can be extrapolated to understand, in terms of days/week/months how much time a Disk on Module will last. Anyone with field experience? Thanks, -- Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de PanamaCel Panama. +(507) 6694-4780 ___ --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 RT on Disk On Module Performance and Durability
I understand Jeremy and Kris point of view (BTW Kris, astlinux rocks!!) However the main question was not aswered (or i didn't get it, did I ?) If I use a Disk on Module that has 2million hours MTBF and a Read/Write lifecycle of 2million times, then, How many days/weeks/months/years will take to do 2million read/write cycles? which leads to my second question. How do I measure/count the read and writes a normal linux system running asterisk does during a day, so I can extrapolate that in terms of time? Is there an utility? Example: if I setup system XYZ with asterisk, then load this magical utility/procedure that counts how many writes the filesystem has done to / or to /,/tmp,/var and after 24 hours the utility/procedure says: 10thousand writes, then, I will do 10thousand writes a day multiplied by200 days= 2 millions Obviously this means I will not use a RAM disk and I want to write to the module everytime Then i will assume that the Disk on a Module will die after 200 days. Or am I completely and horribly misunderstanding the 2million Read/WriteLifeCyleadvertised by Disk-on-Module companies? Example: http://www.pqi.com.tw/product2.asp?oid=140cate1=143PROID=34 ‧MTBF:2,000,000 Hours‧R/W Cycle:2,000,000 Times I want to understand if that's what they mean. I fully understand that such media will have a longer life cycle if i only read from it and keep writes to a mimimum, for example: writing dialpan changes. The whole idea comes from doing a mini itx with no moving parts offering voicemail stored in a disk-on-module and astlinux in a CF and a RAM Disk large enough to do processing on RAM before saving to CF or to disk-on-module when needed. Thanks again for you comments, On 10/6/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Kristian Kielhofner wrote: Erick, OrJust use AstLinux which kind of does what Jeremy described :) http://www.astlinux.org P.S. - I am the creator of AstLinux -- Kristian Kielhofner Sorry to reply to my own post, but there seems to have been some confusion in what I said here.To completely clear it up, Astlinux onlywrites to flash in these circumstances:1)You update the configs.2)You update AstLinux.3)You are using voicemail and people leave voicemail. (most flash seems to last long enough given typical voicemail usage patterns)4)If you have the PERSISTLOG option enabled, I will save syslogs toflash (not RAM - the default).Users are warned about this, and it is not the default.5)astdb is stored in flash, so depending on your needs, SIPregistrations and/or dundi keys may get written here periodically.Imight make an option similar to PERSISTLOG to disable this. Also, you have the option of using a hard drive or alternate flashdevice for ALL writes.Boot from flash, run from HD.Do whatever worksbest for you and your application.--Kristian Kielhofner ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de PanamaCel Panama. +(507) 6694-4780 ___ --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 RT on Disk On Module Performance and Durability
Jeremy, Cohen, Kris, thanks to all of you. Indeed after reading the Sandisk paper it shed a lot of light on this matter. The whole idea is to have a large scale system with no moving parts (we call a large system something with250 users, at least down here ;-) ) the whole idea is for a customer that needs an IVR in 4 languages with autoattendant, extensive CDR and plotted usage patterns as well as voicemail. Voicemail will be used *a lot*, probably about one thousand voicemails per day and the customer does not want VM-to-Email (God knows why!). Oh, and the whole idea of the database is because the developers are working in an AJAX based interface that does the asterisk config/plotting/vm/day-to-day stuff with ARA, so a db is needed. I started learning asterisk with flat files...it works for me...but hey...times are changing. Who knows, maybe the whole thing can be fitted in ram (except for the vm part)...we'll see. I had to ask anyway, but i don't like Dbs eitherit adds and extra breakup layer (maybe Im kind of outdated). Smaller iPBXs will definitely be CF and RAM based and I, at least, will force VMtoEmail and do all the processing in RAM. Again, Thanks to all of you. P.D. I will later follow this thread with the full working configs that will take place at user premises. And for the sake of the test. I will try to kill a sandisk USB with the full config. On 10/8/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Jeremy McNamara wrote: Tzafrir Cohen wrote: H, I'm not sure that this is exactly the data you're after. You're looking for the ammounts of writes for the disk block that gets the most writes. E.g: for a standard ext3 filesystem, the journal area would probably have very frequent writes, whereas most of the system would remain mostly unchanged. Again, if the embedded system is setup properly, there is NO writing to the flash during normal operations, thus the device won't be killed by its alleged 2 million write limitation. Kris and I had a quick discussion on this topic, off-list, and his original flash-based device is still in constant operation after 2 years and I have flash modules that I purposely tried to kill with writes. It took significant effort to start causing error situations, which were very easily detected before the system would become unusable. Erick, you should focus on having a quick action restoration plan and extra DOMs always readily available.Then when a failure situation is detected, you can react very quickly. Jeremy McNamaraJeremy, Erick - I have always pointed to this SanDisk whitepaper: http://www.sandisk.com/Assets/File/OEM/WhitePapersAndBrochures/RS-MMC/WPaperWearLevelv1.0.pdf While it specifically discusses their industrial line of CF cards, it is pretty obvious that flash can, and often does, last much longer thanother components in a system when properly implemented.You will noticethat the SHORTEST expected life of a CF card in their test scenarios was over 70 years!How long is your power supply going to last?Even ifthe consumer level cards had 1/10 the life expectancy, that is stillseven years.I expect to get at least that from my original AstLinux system.It's been two so far, I'll let you know how it is doing inanother five years :). JFFS (and similar FSs) are not appropriate for CF cards or DOMs.Theyare meant to be used directly on flash memory and do their own wear leveling and in some cases, compression.All kinds of commercialdevices use JFFS2.If you are using a CF or DOM with Linux, ext2 is thebest FS to use.CF cards and DOMs use their own wear leveling, so none is required in the operating system or file system.CF cards and DOMshide wear leveling from you and expose themselves as an ordinary IDE device. I echo Jeremy's conclusions.With a properly designed operating system, decent flash memory, and a reasonable usage pattern, I can tellyou (with a great amount of certainty) that in most situations, CF cardswill outlast just about any hard drive (even SCSI) when used 24/7. These days, it really is pretty tough to trash flash. However, if you are running a MySQL cluster or something with several,multi-gigabyte databases, no type of flash memory will last very long! :) To get back to answering your question, I HIGHLY recommend that youavoid MySQL and realtime on your box with a DOM.Nothing against either(MySQL or Realtime), but they will probably make your device more complicated than it needs to be while substantially shortening the lifeof your DOM.If you absolutely have to use MySQL, you might have betterluck using a MySQL storage engine that uses fewer writes than InnoDB, but I am no expert on that.--Kristian Kielhofner___--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 -- Erick PerezPanama
Re: [asterisk-users] Asterisk RT on Disk On Module Performance andDurability
Douglas, Im just the asterisk guy. If they decide to write a cross-browser multi-tier interface in AJAX, assembly language or Pascal, that's up to them (the programmers). I will let them know what can/can't be done. Thinking of that...15 years ago...the last time i used pascal. On 10/9/06, Douglas Garstang [EMAIL PROTECTED] wrote: I'm just going to jump in here, and ask a stoopid question. How could you possibly write a multi-user front end in AJAXwithout using a database backend like MySQL? Doug. -Original Message-From: Erick Perez [mailto: [EMAIL PROTECTED]]Sent: Monday, October 09, 2006 1:58 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk RT on Disk On Module Performance andDurability Jeremy, Cohen, Kris, thanks to all of you. Indeed after reading the Sandisk paper it shed a lot of light on this matter. The whole idea is to have a large scale system with no moving parts (we call a large system something with250 users, at least down here ;-) ) the whole idea is for a customer that needs an IVR in 4 languages with autoattendant, extensive CDR and plotted usage patterns as well as voicemail. Voicemail will be used *a lot*, probably about one thousand voicemails per day and the customer does not want VM-to-Email (God knows why!). Oh, and the whole idea of the database is because the developers are working in an AJAX based interface that does the asterisk config/plotting/vm/day-to-day stuff with ARA, so a db is needed. I started learning asterisk with flat files...it works for me...but hey...times are changing. Who knows, maybe the whole thing can be fitted in ram (except for the vm part)...we'll see. I had to ask anyway, but i don't like Dbs eitherit adds and extra breakup layer (maybe Im kind of outdated). Smaller iPBXs will definitely be CF and RAM based and I, at least, will force VMtoEmail and do all the processing in RAM. Again, Thanks to all of you. P.D. I will later follow this thread with the full working configs that will take place at user premises. And for the sake of the test. I will try to kill a sandisk USB with the full config. On 10/8/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Jeremy McNamara wrote: Tzafrir Cohen wrote: H, I'm not sure that this is exactly the data you're after. You're looking for the ammounts of writes for the disk block that gets the most writes. E.g: for a standard ext3 filesystem, the journal area would probably have very frequent writes, whereas most of the system would remain mostly unchanged. Again, if the embedded system is setup properly, there is NO writing to the flash during normal operations, thus the device won't be killed by its alleged 2 million write limitation. Kris and I had a quick discussion on this topic, off-list, and his original flash-based device is still in constant operation after 2 years and I have flash modules that I purposely tried to kill with writes. It took significant effort to start causing error situations, which were very easily detected before the system would become unusable. Erick, you should focus on having a quick action restoration plan and extra DOMs always readily available.Then when a failure situation is detected, you can react very quickly. Jeremy McNamaraJeremy, Erick - I have always pointed to this SanDisk whitepaper: http://www.sandisk.com/Assets/File/OEM/WhitePapersAndBrochures/RS-MMC/WPaperWearLevelv1.0.pdf While it specifically discusses their industrial line of CF cards, it is pretty obvious that flash can, and often does, last much longer thanother components in a system when properly implemented.You will notice that the SHORTEST expected life of a CF card in their test scenarios was over 70 years!How long is your power supply going to last?Even ifthe consumer level cards had 1/10 the life expectancy, that is still seven years.I expect to get at least that from my original AstLinuxsystem.It's been two so far, I'll let you know how it is doing inanother five years :). JFFS (and similar FSs) are not appropriate for CF cards or DOMs.They are meant to be used directly on flash memory and do their own wear leveling and in some cases, compression.All kinds of commercialdevices use JFFS2.If you are using a CF or DOM with Linux, ext2 is the best FS to use.CF cards and DOMs use their own wear leveling, so noneis required in the operating system or file system.CF cards and DOMshide wear leveling from you and expose themselves as an ordinary IDE device. I echo Jeremy's conclusions.With a properly designed operating system, decent flash memory, and a reasonable usage pattern, I can tellyou (with a great amount of certainty) that in most situations, CF cards will outlast just about any hard drive (even SCSI) when used 24/7. These days, it really is pretty tough to trash flash. However, if you are running a MySQL cluster or something with several,multi-gigabyte databases, no type of flash memory will last very long
[asterisk-users] OT: BioFuel to power phone networks
This are the things that make me believe in technology. I wonder if Ubuntu Linux advocates will help with the development of the controlling modules. * Reuters 16:55 PM Oct, 11, 2006 AMSTERDAM -- Palm and pumpkin seed oil could soon be generating electricity to help power cell phone networks across Africa under a plan to replace fossil fuels with sustainable biofuels made from crops grown by local farmers. Full Story: http://www.wired.com/news/wireservice/0,71936-0.html?tw=rss.technology -- Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Issues with Asterisk 1.4 Beta
My 2 cents but im still playing with 1.4 Issue 5: on the phones disable silence supression or set to yes the transmit silence option. I am not sure if that is the nameof the option in Swiss phones but the whole idea is to *not* save bandwidth when the line goes silent (because both sides stop talking). Make sure ALL SIP phones have disabled silence suppression you may as well take a look at: bug 5374, which allows Asterisk to communicate with devices that support silence suppresion; bug 5409, comfort noise generation in Asterisk; and bug 1234. cheers, On 10/12/06, Jason Walker [EMAIL PROTECTED] wrote: I thought I would list my issues so all of you that know more than me might be able to help. 1. I have 6 Swissphone ip10 they disconnect calls at either 70 seconds, 120 seconds or 180 seconds I have polycom Phones that go forever 2. When I try and transfer calls I have a LONG delay before the seconds line is usable. Call1 on hold then make second call and 1 minute passes before it attempts a connect 3. I have many Polycom 501s and I cannot seem to get the tick server to work. I change settings but it does nto fetch the time 4.I get-- Got SIP response 500 Internal Server Error back from 192.168.0.XXX from all my Polycom 501 phone every 2 mintues or so 5. I get [Oct 12 08:49:56] NOTICE[29165]: rtp.c:708 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.0.141 on my Swiss phones Any help would be great. I am a little new to asterisk and so if I posted this incorrectly please let me know Jason Walker ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] 1.2.12.1 crashing
Maybe a total dumb question but I see you talk about the 1.0.x version and the 1.2.x version. I always see references to the 1.2.x version. Where can I read about the differences in 1.0 and 1.2? Isn't the 1.0 version only available when you buy ABE ? On 10/13/06, Joseph [EMAIL PROTECTED] wrote: On Fri, 2006-10-13 at 07:27 +0200, Remco Barendse wrote: On Thu, 12 Oct 2006, Eric ManxPower Wieling wrote: Matt Florell wrote: If you downgrade, let us know if it fixes things for you. It's strange that there were so many changes in the 1.2 SVN branch after 1.2.7.1 that seem to be complete changes in how some things operate(like the transcoding optimization mess for Asterisk 1.2.11 and 1.2.12 that was fixed in 1.2.12.1). I wish that such radical changes would not be made in a release branch at the expense of reliabitily. Maybe Digium can run the next release for 7 days on their PRODUCTION Asterisk box before a release. I guess they did, and it probably worked. Then they run it for several months, and if it works they label it Business Edition and actually sell it because they know it will work. What hardware are they testing it with, just Digium cards? Asterisk 1.2.12.1 definitely doesn't run correctly with Sipura 3000, as it crashes on second call to PSTN line. -- #Joseph ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Electric usage of a tdm400p
Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? -- Erick Perez ___ --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] Electric usage of a tdm400p
Well Im planning to use a mini-itx, a laptop hdd and a 4fxs digium card. the mini-itx comes with a 60W DC to DC adapter (80W peak). So I need power to manage the hdd, motherboard,the tdm card. A disk cable can be made available, but is not present as a factory default. So My real concern is power. On 10/18/06, Bob Chiodini [EMAIL PROTECTED] wrote: On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote: Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? Erick, Per http://en.wikipedia.org/wiki/Ring_(telephone) in the US the ring voltage is around 90VAC (20 Hz) with a current of 30 milliamperes (REN ~5). This translates to 2.7 watts. Assuming a DC/DC converter efficiency of 38% (probably low), you would need about 3.7 watts, per FXS module. About 15 watts, total. What is the TDM card installed in and is a disk drive cable available? Bob... ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 to decrease Asterisk load
sangoma? voicetronix? they have builtin dsp. they support asterisk. On 5/12/05, Mamadou Lamine KA [EMAIL PROTECTED] wrote: Thanks Mike, I am already using rawplayer for music-on-hold. I have been told of IpVolution TDM60 card that has DSP resources ... Does someone out there ever experienced it? Lamine - Original Message - From: Mike Holloway [EMAIL PROTECTED] To: Mamadou Lamine KA [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 12, 2005 6:20 PM Subject: Re: [Asterisk-Users] How to decrease Asterisk load One thing I do is use rawplayer instead of mpg123 for music-on-hold playback, so that mp3's don't have to be decompressed in realtime. See the wiki for details on using sox to convert your audio samples to raw format, and how to configure musiconhold.conf to use rawplayer to play these files. -mike Mamadou Lamine KA wrote: Hi everybody, I would like to decrease the load of my asterisk server. Could someone recommend me a solution? I have thought about a hardware component that would do some tasks as compression/decompression or codec translations but wonder if such a solution exist. Thanks for any suggestion Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Something every TDMP user should know
I have never had to play with setpci before. Can you elaborate on the use and purpose of this command? On 5/12/05, Colin Anderson [EMAIL PROTECTED] wrote: They instantly got us to look at the output of zttest and we found that this was (in their words) 'extremely low', with 'best' and 'worst' readings of 99.975586% and 99.963379% respectively. Might want to give PCI latency setting a try, it helped for me. My ZTTEST would drop occasionally to 99.95% until I set: setpci -v -s 01:01.0 latency_timer=ff --Digium PRI card setpci -v -s 01:04:0 latency_timer=ff --Digium 401 4 X FXS setpci -v -s XX:XX:X latency_timer=0 --1 entry for every other PCI card in system from LSPCI output, modify XX:XX accordingly Before setpci I would get best in ZTTEST at 99.987793% and worst ~ 99.95% After setpci best is 100% and worst is 99.987793% consitient. I use SpanDSP to recieve faxes and before faxes were garbled and now they are OK (BTW, now recieving ~150 faxes a day 99.95% OK, so SpanDSP *does* work fine, you just have to set it up right. Ask me how.) I put the setpci statements in /etc/rc.d/rc.local before my modprobes to the Digium hardware and Asterisk startup. I'm using a 4-way Netfinity FC2 * 1.0 stable I dunno, maybe the community is being too hard on Digium about the design of the card. I can understand their perpective, it's brutal to make a card that has to have such tight tolerances and make it work acceptably on the huge variation in white box hardware (or black box, in your case). There's a page on the Wiki about motherboards that work well with installation notes but that's pointless since motherboards are such a moving target. Even the motherboard vendor screwing around with BIOS updates can invalidate that information. What I think is best for Asterisk implementation is for Digium to sell a motherboard. No, seriously. Find a ECS or Abit or ASUS mobo that consitiently yields 100% or 99.% and white-box it as a barebones kit with a TXXX card. Sell it as a case, good PSU, mobo, and TXXX card - you add your own RAM, NIC, CPU HDD. Would you buy one for $699? I probably would. It took me a couple of months of fooling around with my Netfinity before I was pleased with the performance and satisfied that it would handle the things I wanted it to do without choking. If I had the option of saving the couple of months time obsessing over things like timing for $699, it would have been a no brainer. Digium wins too, because they get an incremental sale that they can make money on (margin on the mobo) and lower support costs because they don't have to chase down IRQ latency phantoms. hth my 2c ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to decrease Asterisk load
there was a document somewhere that stated those hardwares had DSP. However I do not use them, so i just read about it a long time ago. Maybe a quick google can shed some light On 5/12/05, Wiley Siler [EMAIL PROTECTED] wrote: neither of those has DSP currently. Sangoma reportedly has fewer IRQ problems. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gustavo Alvarez Sent: Thursday, May 12, 2005 2:29 PM To: Erick Perez; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to decrease Asterisk load digium does not have builtin dsp?? is sangoma better than digium?? Erick Perez escribió: sangoma? voicetronix? they have builtin dsp. they support asterisk. On 5/12/05, Mamadou Lamine KA [EMAIL PROTECTED] wrote: Thanks Mike, I am already using rawplayer for music-on-hold. I have been told of IpVolution TDM60 card that has DSP resources ... Does someone out there ever experienced it? Lamine - Original Message - From: Mike Holloway [EMAIL PROTECTED] To: Mamadou Lamine KA [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 12, 2005 6:20 PM Subject: Re: [Asterisk-Users] How to decrease Asterisk load One thing I do is use rawplayer instead of mpg123 for music-on-hold playback, so that mp3's don't have to be decompressed in realtime. See the wiki for details on using sox to convert your audio samples to raw format, and how to configure musiconhold.conf to use rawplayer to play these files. -mike Mamadou Lamine KA wrote: Hi everybody, I would like to decrease the load of my asterisk server. Could someone recommend me a solution? I have thought about a hardware component that would do some tasks as compression/decompression or codec translations but wonder if such a solution exist. Thanks for any suggestion Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream ATA286 outgoing voice
Hi there, Being somehow new to this I like to be provided with guidance as to how to diagnose a potential problem/bottleneck with my GS ata286. my internet speed is 512kbps downstream with 128 kbps upstream with a local cablemodem provider. While i can make and receive calls perfectly, my calls cannot last more than 10-20 minutes without start losing the conversation. Inboud voice sounds perfectly every time, even when i have problems. outbound voice is the problem, people cannot hear me or hear me distorted AFTER 10-20 minutes of crystal-clear talk. Note: all call are from my ata286 voip to land lines. or land lines calling me. codec: g729b firmware:??? can someone point me to the latest firmware, if any? thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 and latency measures
Hi, we have set up a small project in a school the following way: SITE_A(4 port analog to ip g729)--ADSL_ISP1---ISP2Asterisk-PSTN Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit) The asterisk box gets internet service via a wireless antenna. 1 Mbit of up/down bandwith Comments: So far, this means that I will need licenses for the 729. asterisk only supports 20ms sampling on g729 so 4 channels will need 96 kilobits at 20ms sampling (or is it kilobytes??) for the internet bandwith. i cannot use CRTP because i cant be sure if the ISP's routers are CRTP aware. Installing ADSL from ISP1 on the asterisk place will give a clear advantage Please correct any of my prior statements if wrong. should I maintain packet latency below 300ms or 150ms? How can I measure this latency all the way to the asterisk? Should I ping from SITE_A to the asterisk box with 8k packets? If I can't install ADSL for the moment, will the above setup work? thanks in advance for all your help. Erick. ___ --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 and latency measures
Thanks Rich, but i'm only allowed to use g729. you said that some folks run high latency connections, but is 300ms high in my setup? On 3/19/06, Rich Adamson [EMAIL PROTECTED] wrote: Erick Perez wrote: Hi, we have set up a small project in a school the following way: SITE_A(4 port analog to ip g729)--ADSL_ISP1---ISP2Asterisk-PSTN Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit) The asterisk box gets internet service via a wireless antenna. 1 Mbit of up/down bandwith Comments: So far, this means that I will need licenses for the 729. asterisk only supports 20ms sampling on g729 so 4 channels will need 96 kilobits at 20ms sampling (or is it kilobytes??) for the internet bandwith. i cannot use CRTP because i cant be sure if the ISP's routers are CRTP aware. Installing ADSL from ISP1 on the asterisk place will give a clear advantage Please correct any of my prior statements if wrong. should I maintain packet latency below 300ms or 150ms? The objective should be to keep latency as low as possible, however some folks do run asterisk via satellite which as a very lengthy latency. How can I measure this latency all the way to the asterisk? Several ways depending on how accurate a measurement you want. A simple ping would give a starting point. A much more expensive way is to use VoIP analysis software to measure it, but be prepared to spend at least $1,500 (US) to do that. Should I ping from SITE_A to the asterisk box with 8k packets? If you want to emulate a sip/iax packet, use a packet size of about 200 bytes. If I can't install ADSL for the moment, will the above setup work? Probably a bigger issue to address relates to what other traffic might be passing across the dsl and/or wireless channel that might be consuming bandwidth and impacting the rtp packets. Broadcasts originating from devices outside your control (other isp users), hackers attempting to access your ip addresses (at both ends), data traffic between your two endpoints, etc, are just some thoughts of items using a portion of the bandwidth available. Might also think about jitter (eg, variations in latency) and what that might do to your end to end communications. There are other low bandwidth codecs available that could be used instead of g729. Some include ilbc, g726, gsm, etc. Each consumes different bandwidths, and each provide a slightly different quality of audio. See the wiki for more detail on what each consumes for bandwidth on the wire. ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] welltech Wellgate 3804 in SIP mode
Hi, does anybody have a working config or tips to connect the welltech wellgate 3804 (4fxo) unit to asterisk via SIP ?I think I register it via SIP with my * box, but when sending calls from * to the wellgate the unit does not pass the call to any of the fxo ports. thanks in advance,-- ---Erick ___ --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] welltech Wellgate 3804 in SIP mode
Martin, i guess im in dumb mode today because i don't get what you say, may also be because this will be my first welltech to configure.what im trying to do is:remote_voip_gatewayasterisk---fox/fxs/and_international_voip_providers call will always go - this way, no incoming calls from the right of the diagram side to asterisk.On 3/23/06, Martin Joseph [EMAIL PROTECTED] wrote:On Mar 22, 2006, at 10:24 PM, Erick Perez wrote: Hi, does anybody have a working config or tips to connect the welltech wellgate 3804 (4fxo) unit to asterisk via SIP ? I think I register it via SIP with my * box, but when sending calls from * to the wellgate the unit does not pass the call to any of the fxo ports.I am using the 3701a, which is a 1 FXO 1 FXS deal.The trick for me was the routing tableor something like that fromthe Web based configuration screen.There I changed the default for the FXO to point to IP, and the IP to default to the FXO.Then I also have the line configuration set so the extension I wantto ring ie 2020 is the hotline for the FXO.For outbound ones, I think I just have a regular old Dial(SIP/[EMAIL PROTECTED] in my dialplan where 2003 is the FXO port.HTH?Marty___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ---Erick PerezLinux User 376588http://counter.li.org/(Get counted!!!)Panama, Republic of Panama ___ --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[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing
Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel) no hardware interfaces installed gives me this error. Im a bit new to this so any help will be appreciated. == Parsing '/etc/asterisk/musiconhold.conf': Found Mar 26 00:58:49 WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing... Sound may be choppy.[chan_oss.so] = (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found == Registered channel type 'Console' (OSS Console Channel Driver)musiconhold.conf has:[default]mode=quietmp3directory=/var/lib/asterisk/mohmp3thanks,-- ---Erick ___ --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] WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing
why should I? i thought in 2.6 kerneles that was not necesary when you dont have physical internfaces on the system.On 3/26/06, Jonathan Augenstine [EMAIL PROTECTED] wrote: Have you verified that ztdummy is loaded?On Sun, 2006-03-26 at 01:06 -0500, Erick Perez wrote: Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel)no hardware interfaces installed gives me this error. Im a bit new to this so any help will be appreciated. == Parsing '/etc/asterisk/musiconhold.conf': Found Mar 26 00:58:49 WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing...Sound may be choppy. [chan_oss.so] = (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found == Registered channel type 'Console' (OSS Console Channel Driver) musiconhold.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 thanks, -- --- Erick ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ---Erick PerezLinux User 376588http://counter.li.org/(Get counted!!!) Panama, Republic of Panama ___ --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
