Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Steve Totaro wrote: I have consulted on so many systems with poor audio, the first thing I check is IAX or SIP. If IAX, I move over to SIP and the calls are prefect. I avoid IAX at all costs, use OpenVPN, open tons of ports on your firewall, whatever you can do to use SIP. The only time I will use IAX is if in some remote backwards part of the world, they have several NATs so it is impossible to control. Even www.iax.cc recommends using SIP. Overhead is of little concern with MPLS and big pipes. I would be interested to hear if you have had any of these problems with the latest 1.4 versions of Asterisk. A _lot_ of work has gone into IAX2 support in Asterisk 1.4, and specifically, the most recent 25% of the 1.4 series or so. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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 1.6 vs 1.4?
What model in the Polycom or Aastra range is the 360 level with? 2008/6/6 Chris Bagnall [EMAIL PROTECTED]: When I pushed some vendors for prices there was only a tiny gap between the 300 and 360. Would suggest looking hard at the 360 always... Interesting... here in the UK the price difference between the 300 and 360 is pretty huge. Either you're getting some stunningly good pricing on 360s or some abysmal pricing on the 300s :-) Regards, Chris ___ -- 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 -- http://www.suretecsystems.com/services/openldap/ ___ -- 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 1.6 vs 1.4?
2008/6/7 Gavin Henry [EMAIL PROTECTED]: What model in the Polycom or Aastra range is the 360 level with? Probably the IP601: http://www.voipon.co.uk/polycom-soundpoint-ip601-p-121.html and 57i: http://www.voipon.co.uk/aastra-57i-ip-phone-p-420.html Snom 360: http://www.voipon.co.uk/snom-360-ip-telephone-p-31.html They all have the 12 keys. ___ -- 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 1.6 vs 1.4?
On Thu, Jun 5, 2008 at 4:45 PM, Johansson Olle E [EMAIL PROTECTED] wrote: 5 jun 2008 kl. 20.45 skrev Michael Graves: I wonder why more vendors haven't adopted IAX yet? I expect that before major players adopt this protocol it'd need to be confirmed as a standard by some form of international body. That was underway, but lacking anyone to push the process along. Please note that the IAX draft is just an informational RFC, not anything that goes any IETF standards track or is endorsed by the IETF. There are many vendor-related protocols documented like that. (Said from the chan_sip corner). Cheers, /Olle I have consulted on so many systems with poor audio, the first thing I check is IAX or SIP. If IAX, I move over to SIP and the calls are prefect. I avoid IAX at all costs, use OpenVPN, open tons of ports on your firewall, whatever you can do to use SIP. The only time I will use IAX is if in some remote backwards part of the world, they have several NATs so it is impossible to control. Even www.iax.cc recommends using SIP. Overhead is of little concern with MPLS and big pipes. Thanks, Steve Totaro ___ -- 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 1.6 vs 1.4?
When I pushed some vendors for prices there was only a tiny gap between the 300 and 360. Would suggest looking hard at the 360 always... Interesting... here in the UK the price difference between the 300 and 360 is pretty huge. Either you're getting some stunningly good pricing on 360s or some abysmal pricing on the 300s :-) Regards, Chris ___ -- 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 1.6 vs 1.4?
Tzafrir Cohen wrote: On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote: Why on earth are you running two layers of echo cancellation - hardware and software? To be honest, I think this is asking for trouble - I've seen two occasions where having Oslec and hardware echo cancellation has caused significant problems with audio quality - the usual symptoms are gaps in the conversation as the hardware cancellation eliminates the majority of the echo and the software cancellation subsequently eliminates parts of the conversation. If you use a hardware EC (or technically: a span-specific echo cancellation method) the generic Zaptel echo canceller (software-based, OSLEC in this case) will not be used. That's not always been my experience with OSLEC. HPEC and the generic Zaptel echo canceller seem to work this way, but as I've said, I've had two cases where I've had to remove OSLEC to stop it degrading voice quality where there is a hardware echo canceller in play. ___ -- 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 1.6 vs 1.4?
On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote: Tzafrir Cohen wrote: On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote: Why on earth are you running two layers of echo cancellation - hardware and software? To be honest, I think this is asking for trouble - I've seen two occasions where having Oslec and hardware echo cancellation has caused significant problems with audio quality - the usual symptoms are gaps in the conversation as the hardware cancellation eliminates the majority of the echo and the software cancellation subsequently eliminates parts of the conversation. If you use a hardware EC (or technically: a span-specific echo cancellation method) the generic Zaptel echo canceller (software-based, OSLEC in this case) will not be used. That's not always been my experience with OSLEC. HPEC and the generic Zaptel echo canceller seem to work this way, but as I've said, I've had two cases where I've had to remove OSLEC to stop it degrading voice quality where there is a hardware echo canceller in play. Unless there are some really strange OSLEC patches floating around, of which I'm not aware, this should not be the case. OSLEC uses exactly the same EC interface to Asterisk as the built-in one and HPEC do. Any chance you didn't actually fully unload zaptel? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Tzafrir Cohen wrote: On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote: If you use a hardware EC (or technically: a span-specific echo cancellation method) the generic Zaptel echo canceller (software-based, OSLEC in this case) will not be used. That's not always been my experience with OSLEC. HPEC and the generic Zaptel echo canceller seem to work this way, but as I've said, I've had two cases where I've had to remove OSLEC to stop it degrading voice quality where there is a hardware echo canceller in play. Unless there are some really strange OSLEC patches floating around, of which I'm not aware, this should not be the case. OSLEC uses exactly the same EC interface to Asterisk as the built-in one and HPEC do. Any chance you didn't actually fully unload zaptel? These weren't installs that we'd done - in both instances, they were companies who had bought Trixbox based systems from someone else who had subsequently gone out of business. Granted, I had /assumed/ that OSLEC was causing problems by not disabling itself properly when hardware echo cancellation was found - and I guess that assumption stuck when the removal of OSLEC resolved the problem. ___ -- 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 1.6 vs 1.4?
On Thu, Jun 05, 2008 at 09:28:52PM +1000, Rob Hillis wrote: Tzafrir Cohen wrote: On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote: If you use a hardware EC (or technically: a span-specific echo cancellation method) the generic Zaptel echo canceller (software-based, OSLEC in this case) will not be used. That's not always been my experience with OSLEC. HPEC and the generic Zaptel echo canceller seem to work this way, but as I've said, I've had two cases where I've had to remove OSLEC to stop it degrading voice quality where there is a hardware echo canceller in play. Unless there are some really strange OSLEC patches floating around, of which I'm not aware, this should not be the case. OSLEC uses exactly the same EC interface to Asterisk as the built-in one and HPEC do. Any chance you didn't actually fully unload zaptel? These weren't installs that we'd done - in both instances, they were companies who had bought Trixbox based systems from someone else who had subsequently gone out of business. Granted, I had /assumed/ that OSLEC was causing problems by not disabling itself properly when hardware echo cancellation was found - and I guess that assumption stuck when the removal of OSLEC resolved the problem. Trixbox has (had?) an earlier version of OSLEC that failed to properly handle the echo training function. And it also sets echotraining=800 (which is probably is not such a good idea anyway nowadays). That combination causes bad audio. Disable echo traning and/or upgrade OSLEC. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Brent, hope your problems go away soon. I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We are using asterisk 1.4.2 for a SIP only based configuration. Currently we have about 200 SIP users which can cause approximately upto 3 simultaneous calls. We are mainly concerned about the performance and stability of asterisk when the load increases. Our server can handle about 100 simultaneous calls having 3Ghz Dual Intl-Xeon Processor with 2GB of ram using G711 codec, and around 30 simultanoeus calls using G729 codec. This we are expecting from the hardware. We are planning to accomodate about 5,000 users on this server. Before the release of 1.6 i heard that its architecture is going to be different from 1.2 and 1.4. Recently i read an article about freeswitchhttp://freeswitch.org/node/117, which explains how its functionality is like asterisk but it can perform better than asterisk due to its architectural differences. The main developer for freeswitch is anthony who also codes for asterisk. He explaines why the architecture of asterisk needs to be changed which requires massive recoding, but nobody took the step to do it. Thats why he started freeswitch on his own to redifine the architecture, so that the performance and reliability of the switch should be better than asterisk. In his article he has already said that freeswitch beats asterisk by a factor of 10. If asterisk architecture is being rewritten in 1.6 to achive the same goal, then we will be happy to use 1.6 instead of shifting the whole system to freeswitch. We dont have any problem or issues with 1.4.2 yet. We are mainly concerned about the its performance when the load increases. If 1.6 is more reliable under heavy loads then we would like to use it. If anyone can put some light on this topic, all i can say is thanx for sharing your thaughts and experiences. On Thu, Jun 5, 2008 at 1:13 AM, Brent Davidson [EMAIL PROTECTED] wrote: Just an update. I tried updating to the newest Rhino Release firmware 1.15 and newest stable driver version 2.2.6. It works OK with zaptel-1.4.9.2 and compiles OK with 1.4.10.1 but when compiled against zaptel 1.4.10.1 Asterisk does not see any zap channels. I'm currently running one branch office with the upgraded firmware, driver, zaptel-1.4.9.2 and Asterisk-1.4.20.1. I'll see how everything goes there and may upgrade the other offices if it works OK. Thanks, Brent ___ -- 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 -- Best Regards Rizwan Hisham ___ -- 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 1.6 vs 1.4?
Hi! I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We are using asterisk 1.4.2 for a SIP only based configuration. [...] We are planning to accomodate about 5,000 users on this server. Many people on this list will advise you to use a SIP proxy like OpenSER in front of Asterisk to take care of SIP registrations - I'll do the same. ;- Before the release of 1.6 i heard that its architecture is going to be different from 1.2 and 1.4. Recently i read an article about freeswitch, which explains how its functionality is like asterisk but it can perform better than asterisk due to its architectural differences. [...] If asterisk architecture is being rewritten in 1.6 to achive the same goal, then we will be happy to use 1.6 instead of shifting the whole system to freeswitch. Looks like you've just decided to move to FreeSwitch ;- While early testing shows that 1.6 can increase SIP performance compared to 1.4 up to factor 3-4, this release is by no means a from-the-ground-up re-write like FreeSwitch. Read more: http://www.voip-info.org/wiki/view/Asterisk+v1.6 http://www.voip-info.org/wiki/view/Asterisk+dimensioning Cheers, Philipp ___ -- 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 1.6 vs 1.4?
Philipp von Klitzing wrote: Hi! I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We are using asterisk 1.4.2 for a SIP only based configuration. [...] We are planning to accomodate about 5,000 users on this server. Many people on this list will advise you to use a SIP proxy like OpenSER in front of Asterisk to take care of SIP registrations - I'll do the same. ;- I've seen comments similar to this going around a lot and I've never really understood it. I guess maybe I won't understand it until I am in a situation where I need to handle a huge call volume and hundreds or thousands of users (I.E. probably never). In my situation, using Asterisk as a distributed PBX with Snom SIP phones I haven't had any problems at all with the asterisk end of the system. All of my problems, and I have to stress they are MINOR problems have been related to interfacing to the analog PSTN. I have not not meshed all of the branch offices together yet, so I may run into further issues there, but all of the inter-office calling will be handled by IAX trunking. I really like these Snom 300 phones as far as audio quality goes. I wish they had a few more programmable buttons but that was a purchasing oversight. We underestimated the number of programmable buttons we would need and opted for the 300 instead of the 360. I also wished they used IAX. It's fairly obvious during the software update process thatthey run either a Linux or BSD derivative so it shouldn't be too difficult to develop an IAX firmware for them, even if it has to be done by a third party. I wonder why more vendors haven't adopted IAX yet? -Brent ___ -- 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 1.6 vs 1.4?
I wonder why more vendors haven't adopted IAX yet? I expect that before major players adopt this protocol it'd need to be confirmed as a standard by some form of international body. That was underway, but lacking anyone to push the process along. I would've thought that Digium would be the most likely lead proponent, but that doesn't seem to be the case. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
On Thu, 2008-06-05 at 13:45 -0500, Michael Graves wrote: I would've thought that Digium would be the most likely lead proponent, but that doesn't seem to be the case. Actually, Digium has been quite active in helping to try to get the IAX protocol adopted as a standard. See http://tools.ietf.org/id/draft-guy-iax-04.txt for the latest draft of the protocol specification as submitted to the IETF. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 1.6 vs 1.4?
5 jun 2008 kl. 20.45 skrev Michael Graves: I wonder why more vendors haven't adopted IAX yet? I expect that before major players adopt this protocol it'd need to be confirmed as a standard by some form of international body. That was underway, but lacking anyone to push the process along. Please note that the IAX draft is just an informational RFC, not anything that goes any IETF standards track or is endorsed by the IETF. There are many vendor-related protocols documented like that. (Said from the chan_sip corner). Cheers, /Olle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Brent Davidson a écrit : ...I wonder why more vendors haven't adopted IAX yet? Well, even ZoIPer (ex IdeFisk) team, still recommend using SIP over IAX as SIP is more mature and reliable in asterisk and zoiper, -- Benoit begin:vcard fn:Benoit Plessis n:Plessis;Benoit email;internet:[EMAIL PROTECTED] tel;home:+33 9 52 49 25 06 tel;cell:+33 6 77 42 78 32 x-mozilla-html:FALSE version:2.1 end:vcard ___ -- 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 1.6 vs 1.4?
Is there some location that outlines the major differences between Asterisk version 1.4 and version 1.6? I've read through change logs and several technical discussions, but technical details don't really give me the big picture. Basically, is 1.6 more stable than 1.4? Is it more efficient? Does it work better with echo cancelers like Oslec? I'm currently using Asterisk as a PBX for our branch offices and will soon be converting our main office. Our goal is to be able to have 2 analog lines at each office, calls come in to each PBX and are routed by VOIP to a receptionist at one of the offices who then routes calls appropriately. We're currently using Asterisk 1.4.19, Zaptel 1.4.10, Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones. All of our branch offices have 1MBPS DSL connections and are linked to each other by VPN's running on our Cisco 1720 routers. Our only problem so far is with intermittent echo on calls. Most of the calls have a little echo right at first, but it goes away almost immediately as the echo canceler trains. Every now and then, however, we get a call with terrible echo. I've put in several e-mails to rhino support asking if the hardware echo canceler needs something I haven't done but didn't get a response. I know echo is just something we have deal with when using analog lines, but I didn't think it would be this big of a problem. All of our offices are in rural areas where digital lines are unavailable so that is not an option. Given this setup, is there any reason for me to switch to Asterisk 1.6 or should I stick with 1.4? Thanks, Brent Davidson ___ -- 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 1.6 vs 1.4?
2008/6/4 Brent Davidson [EMAIL PROTECTED]: [snip] We're currently using Asterisk 1.4.19, Zaptel 1.4.10, Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones. [snip] Just a small aside... You go to the trouble of building/using Oslec, and then use hardware EC? Very odd. Does Oslec understand about not doing EC if the hardware is doing EC? I imagine this is brokered by Zaptel, but am not sure. Perhaps you are doing double-EC and causing breakage? Just a random thought. Steve ___ -- 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 1.6 vs 1.4?
Have you tuned rxgain txgain in Zapata.conf? shameless-self-plug http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/ /plug Also, have you used fxotune to tune each FXO interface? I believe echo cancellation happens at the Zaptel / DAHDI level, so using Asterisk 1.6 probably isn't going to give you any benefit. -- Matt Watson http://www.mattgwatson.ca -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: Wednesday, June 04, 2008 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6 vs 1.4? Is there some location that outlines the major differences between Asterisk version 1.4 and version 1.6? I've read through change logs and several technical discussions, but technical details don't really give me the big picture. Basically, is 1.6 more stable than 1.4? Is it more efficient? Does it work better with echo cancelers like Oslec? I'm currently using Asterisk as a PBX for our branch offices and will soon be converting our main office. Our goal is to be able to have 2 analog lines at each office, calls come in to each PBX and are routed by VOIP to a receptionist at one of the offices who then routes calls appropriately. We're currently using Asterisk 1.4.19, Zaptel 1.4.10, Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones. All of our branch offices have 1MBPS DSL connections and are linked to each other by VPN's running on our Cisco 1720 routers. Our only problem so far is with intermittent echo on calls. Most of the calls have a little echo right at first, but it goes away almost immediately as the echo canceler trains. Every now and then, however, we get a call with terrible echo. I've put in several e-mails to rhino support asking if the hardware echo canceler needs something I haven't done but didn't get a response. I know echo is just something we have deal with when using analog lines, but I didn't think it would be this big of a problem. All of our offices are in rural areas where digital lines are unavailable so that is not an option. Given this setup, is there any reason for me to switch to Asterisk 1.6 or should I stick with 1.4? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Brent Davidson wrote: We're currently using Asterisk 1.4.19, Zaptel 1.4.10, Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones. Why on earth are you running two layers of echo cancellation - hardware and software? To be honest, I think this is asking for trouble - I've seen two occasions where having Oslec and hardware echo cancellation has caused significant problems with audio quality - the usual symptoms are gaps in the conversation as the hardware cancellation eliminates the majority of the echo and the software cancellation subsequently eliminates parts of the conversation. ___ -- 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 1.6 vs 1.4?
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote: Brent Davidson wrote: We're currently using Asterisk 1.4.19, Zaptel 1.4.10, Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones. Why on earth are you running two layers of echo cancellation - hardware and software? To be honest, I think this is asking for trouble - I've seen two occasions where having Oslec and hardware echo cancellation has caused significant problems with audio quality - the usual symptoms are gaps in the conversation as the hardware cancellation eliminates the majority of the echo and the software cancellation subsequently eliminates parts of the conversation. If you use a hardware EC (or technically: a span-specific echo cancellation method) the generic Zaptel echo canceller (software-based, OSLEC in this case) will not be used. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Tzafrir Cohen wrote: On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote: Brent Davidson wrote: We're currently using Asterisk 1.4.19, Zaptel 1.4.10, Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones. Why on earth are you running two layers of echo cancellation - hardware and software? To be honest, I think this is asking for trouble - I've seen two occasions where having Oslec and hardware echo cancellation has caused significant problems with audio quality - the usual symptoms are gaps in the conversation as the hardware cancellation eliminates the majority of the echo and the software cancellation subsequently eliminates parts of the conversation. If you use a hardware EC (or technically: a span-specific echo cancellation method) the generic Zaptel echo canceller (software-based, OSLEC in this case) will not be used. Is there any indication of this in Zaptel?This is the output of my ztcfg -vv: Zaptel Version: 1.4.10 Echo Canceller: Oslec Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels to configure. I removed Oslec when I first installed the R4FXO-EC cards, but echo was terrible and made the calls unusable. My gut instinct is telling me that my hardware echo cancellation is not working. Thanks, Brent ___ -- 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 1.6 vs 1.4?
Matt Watson wrote:
Have you tuned rxgain txgain in Zapata.conf? shameless-self-plug
http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/
/plug
Also, have you used fxotune to tune each FXO interface?
I believe echo cancellation happens at the Zaptel / DAHDI level, so using
Asterisk 1.6 probably isn't going to give you any benefit.
--
Matt Watson
http://www.mattgwatson.ca
FXOTune is apparently not compatible with the R4FXO cards. Here is the
output:
fxotune -i
Tuning module /dev/zap/1
Unable to set impedance on fd 4
Failure!
Tuning module /dev/zap/2
Unable to set impedance on fd 4
Failure!
/dev/zap/3 absent: No such device or address
/dev/zap/4 absent: No such device or address
/dev/zap/5 absent: No such file or directory
/dev/zap/6 absent: No such file or directory
/dev/zap/7 absent: No such file or directory
.
. {multiple lines of same basic message edited out}
.
/dev/zap/246 absent: No such file or directory
/dev/zap/247 absent: No such file or directory
/dev/zap/248 absent: No such file or directory
/dev/zap/249 absent: No such file or directory
/dev/zap/250 absent: No such file or directory
/dev/zap/251 absent: No such file or directory
/dev/zap/252 absent: No such file or directory
Unable to tune 2 devices, even though those devices are present
Thanks,
Brent
___
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Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Just an update. I tried updating to the newest Rhino Release firmware 1.15 and newest stable driver version 2.2.6. It works OK with zaptel-1.4.9.2 and compiles OK with 1.4.10.1 but when compiled against zaptel 1.4.10.1 Asterisk does not see any zap channels. I'm currently running one branch office with the upgraded firmware, driver, zaptel-1.4.9.2 and Asterisk-1.4.20.1. I'll see how everything goes there and may upgrade the other offices if it works OK. Thanks, Brent ___ -- 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
