Re: [Asterisk-Users] H323 to SIP
Farhad Ibragimov wrote: I don’t have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me Asterisk is perfectly documented everywhere on the net. Maybe the first place to visit in order to have working asterisk is www.asterisk.org.Second place is www.voip-info.org If any question arises feel free to email me privately. Tofik Suleymanov ___ --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] more one asterisk hardware
Hello list, We are going to build and deploy telephony-system for approx ~1000 users with ASTERISK as main PBX.I was googling through this mailing list and found a lot of useful information.Some of my questions solved ,some other are still on agenda.Below I will formulate short questions that are still unanswered: 1. What are pros and cons of using SER with ASTERISK ? 2. Should we place SER as a registar server or only as a load balancer (or maybe both) ? 3. What hardware SIP phones are known to work flawlessly in production system with ~1000 users, and where can we buy this amount of phones with the best possible discount ? More about features of those phones please. 4. What hardware (ram,cpu , ... etc etc) should be used in a such environment to make ASTERISK happy ? 5. What if we run ASTERISK on FreeBSD instead of Linux ? Cons and pros please. Thank you, Tofik Suleymanov ___ --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] another question about hardware for using with asterisk
Hello folks, firstly, thank you for your useful and fast answers ! Is there anybody using D-Link SIP phones ? Are D-Link SIP phones ok to install in production environment ? Give your comments please. Tofik Suleymanov ___ --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 hardware
Wilson Pickett wrote: anyone using hardware IAX phones with asterisk ? I've googled on this issue and found several hardware phones which support IAX protocol, but before paying money I'd like to know more about what people experiencing with them. I have had three of them for neary two years. Here's an "executive review": They're dirt cheap You get what you pay for. Some (not all) firmware has clicks and hums. The later phones look and sound better and are more robust. You can get the firmware and tweak it (see yahoo group) Many languages are available since users have the firmware They do both SIP and IAX, (one at a time via firmware change) They will speak the ip address, server address etc, good for unsighted users. Many don't like this and have disabled it (by a firmware tweak) Conclusion: these phones are excellent for use on the road (assuming a real Internet connection and not through a proxy as many hotels do). I wouldn't recommend them for daily intensive use as they aren't built for it. All of my phones have been continuosly online and working for over one year. They are perfect for setting up asterisk, tinkering with firmware and settings, giving a phone to distant firends or relatives. Google for yuxin,atcom, yahoo for PA1688 mailing list and here http://www.voip-info.org/wiki-VOIP+Phones ___ --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 Wilson, thank you very much for extensive and useful answer ! Tofik Suleymanov ___ --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 hardware
Steve Totaro wrote: Give idefisk a try. It works very well for me, its free, and does not crash all the time like Cubix (formerly Firefly). Hello Steve, As far as i know 'idefisk' is a softphone, but i need a hardware phone. Thank you for reply. Tofik Suleymanov ___ --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 hardware
Hello folks, anyone using hardware IAX phones with asterisk ? I've googled on this issue and found several hardware phones which support IAX protocol, but before paying money I'd like to know more about what people experiencing with them. Thank you, Tofik Suleymanov ___ --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 voicemail question
Hello list, When new voicemail comes and i pick up the phone i hear special tones indicating that the new voicemail arrived.I've never had any problems with this feature, but several days ago it begin to behave strangely: 1. new voimcemail arrives, but i dont hear the special indicating tones when picking up the phone 2. there is no new voicemail (checked mailbox on filesystem), but when i pick up the phone i hear speial tones indicating that there is a new message is this a known issue ? ___ --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] transforming g729 files to wav files
Noah Miller wrote: Hi Tofik - is there any open-source software that recodes g729 sound files to wav sound files ? The only way (at least) to do such transformation is with interactive form on: http://www.asteriskguru.com/audio_conversion.php The wiki also lists GX::Transcoder which looks like it can do g729 to wav, though I've never tried it. Here's a link: http://www.germanixsoft.de/index.php Otherwise, you could probably rig up asterisk to transcode from g729 to another codec then record it to a file. There's probably not more tools to do this since most people aren't interested in going from the very lossy g729 codec to the non-lossy wav format. I am aware of this tool, but is there any other tool (same features like mentioned gx::transcoder) for unix-like operating systems ? ___ --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] transforming g729 files to wav files
Darrell Long wrote: The resulting file is not going to sound any better and its going to take up more space. What is the reason you need a WAV file? Perhaps there is a more efficient way to do what you are trying to do. Darrell S. Long BestWeb Corporation I understand issues about sound quality.Here is the situation: i am using g729-native sound files and g729 codecs everywhere.My voicemail is coming in g729 format also.Some time ago one of our customers asked for the voicemail to go to his e-mail and i want him to recieve just a .wav file. I've also tried to use: format=g729|wav in my voicemail.conf in order to have copies of voicemails in wav format but for unknown reason (after this change) i wasnt able to hear voicemail announcements when trying to access voicemail. ___ --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] transforming g729 files to wav files
Hello list, is there any open-source software that recodes g729 sound files to wav sound files ? The only way (at least) to do such transformation is with interactive form on: http://www.asteriskguru.com/audio_conversion.php Tofik Suleymanov ___ --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] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P
Dovid Bender wrote: Can Asterisk serve as an access server/gateway to the internet? I have the same question. If I had a PRI coming in to asterisk can I have users dial in and have asterisk work as a gateway to the internet ? Dovid Very interesting question. if this feature is absent, is it possible to add a module for doing such thing ? And how hard will it be to implement it. Tofik Suleymanov ___ --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] sipura spa2 + asterisk bug ?
Vahan Yerkanian wrote: Tofik Suleymanov wrote: 1. assume 1-st line is in use 2. after dialing 2-nd line from outside i immediately go to the voicemail announcement (also i immediately go to voicemail if i dial from extension to extension both of which are on the same sipura device) Check what response code appears on Asterisk CLI when you dial 2nd line. If your SPA-2k is SPA-2000 or SPA-2002 there might be a codec combination that is still overloading CPU and it's sending back unavailable response. I assume both extensions have separate username/passwords, don't they? Vahan ___ --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 Thank you, Vahan i've managed both of lines to work after playing a bit with codecs. ___ --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] sipura spa2 + asterisk bug ?
Steve Kennedy wrote: On Tue, Mar 28, 2006 at 07:40:06PM +, Tofik Suleymanov wrote: Each of the two lines have their own entry in sip.conf and i can see each line registered in 'sip show peers'. I can dial each line from outside successfully but when one line is busy i can't reach the second line (it immediately sends me to the voicemail).I've also tried to change the timeouts in dial command but seems that it doesn't matter. Any other advice ? You haven't got codec negotiation set-up properly so it's still running out of g.729 and then it will act as busy I have dtmfmode=rfc2833 disallow=all allow=g729 allow=gsm allow=alaw allow=ulaw allow=g723.1 So should try g.729 first, then gsm (which unfortunately SPA don't support), etc etc. Steve Thank you very much ! after playing a bit with codecs i've managed my sipura lines to work properly. Again, thank you very much for quick and effective help ! Tofik Suleymanov ___ --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] sipura spa2 + asterisk bug ?
Vahan Yerkanian wrote: Tofik Suleymanov wrote: Thanks all for replying tommorow i'll try to do like you suggested. One more quick question: why Sipuras cant do more than 1 g.729 channel at a time ? Insufficient CPU power to process 2 g729 streams. Is this somehow related to g.729 licensing ? Is there any other SIP adapters which override this drawback ? You're looking for Sipura SPA-2100. It has more powerful CPU. Check http://www.sipura.com/support/spa2100faq/Section_1.html HTH, Vahan ___ --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 Vahan, thank you for reply. i've changed codec types for my sipura lines and now 2 simultaneous calls from my sipura device to outworld are working, but here is the other problem i came across: 1. assume 1-st line is in use 2. after dialing 2-nd line from outside i immediately go to the voicemail announcement (also i immediately go to voicemail if i dial from extension to extension both of which are on the same sipura device) Any clues on what is happening ? Thank you, Tofik Suleymanov ___ --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] sipura spa2 + asterisk bug ?
Steve Kennedy wrote: On Tue, Mar 28, 2006 at 01:20:06AM +0300, Tofik Suleymanov wrote: How to reproduce this bug (?) : 1. register sipura spa2 with 2 lines on asterisk. 2. use first line to call somewhere. 3. while using first line try to call from second line somewhere else in 3 step i hear fast busy tones on second line and asterisk console gives me this short error: Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No compatible codecs! My sipura adapter is using g729a codec. When using both of sipura lines separately everything works fine, until someone tries to use both lines simultaneously. The Sipuras only support 1 g.729 codec at any time. If one channel is using g.729, the other channel has to use another codec. You'll just have to open more than g.729 at the Asterisk end, so the first channel can use g.729 and the other another codec. Steve Thanks all for replying tommorow i'll try to do like you suggested. One more quick question: why Sipuras cant do more than 1 g.729 channel at a time ? Is this somehow related to g.729 licensing ? Is there any other SIP adapters which override this drawback ? Tofik Suleymanov ___ --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] sipura spa2 + asterisk bug ?
Hello, How to reproduce this bug (?) : 1. register sipura spa2 with 2 lines on asterisk. 2. use first line to call somewhere. 3. while using first line try to call from second line somewhere else in 3 step i hear fast busy tones on second line and asterisk console gives me this short error: Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No compatible codecs! My sipura adapter is using g729a codec. When using both of sipura lines separately everything works fine, until someone tries to use both lines simultaneously. Any advice ? Tofik Suleymanov ___ --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
