Re: [asterisk-users] Had it with Dell Garbage
We currently using the x service servers as well, never had any problems with them. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, 27 February 2008 7:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage On Tue, Feb 26, 2008 at 3:20 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Tue, Feb 26, 2008 at 3:10 PM, Matt [EMAIL PROTECTED] wrote: I've had it with Dell server garbage.They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver.They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I change underwear. [end rant]. Can anyone recommend an IBM or Gateway server that you have used with Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has room for one or two PCI-express interface cards? HP DL380 is my baby. Thanks, Steve Totaro IBM X series are also great, I have deployed many, I just have a thing for HP. 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 ___ -- 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] Call go into a HOLD music instead
What hardware are you running at the moment? Cheers, Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sanjoy Rath Sent: Thursday, 7 February 2008 12:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call go into a HOLD music instead Thanks Doug for your email. I will assign another extension so that it does not comflict with 700. Also I see CPU being 100% used. DO not know if I can stop something to increase the CPU idle %. Thanks, Sanjoy. Date: Wed, 6 Feb 2008 20:35:13 -0500 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call go into a HOLD music instead Sanjoy Rath wrote: I am not dialing ext 701 but 700 from 500 ext. Do not know why its going to 701. Check your features.conf, 700 by default is the parking extensions. It will place calls on 701 first, and then 702 if 701 isn't available. This can be changed via the features.conf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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
[asterisk-users] Source Based Call Routing
Hi List, I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it. What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use specific outbound trunks. Approximately a third of the extensions will one the first trunk, a third the second trunk, and the rest will use the last trunk. We also need control over assigning with trunks the given extensions will use. Any suggestions on how to get this to work would be very much appreciated. Many Thanks, Daniel ___ -- 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] Source Based Call Routing
Thank you Greg and Alex for your contribution. I will use your leads to see what I can get asterisk to do :) Many Thanks, Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grey Man Sent: Wednesday, 30 January 2008 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Source Based Call Routing - Original Message From: Daniel Cole [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 10:31:55 PM Subject: [asterisk-users] Source Based Call Routing Hi List, I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it. What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use specific outbound trunks. Approximately a third of the extensions will one the first trunk, a third the second trunk, and the rest will use the last trunk. We also need control over assigning with trunks the given extensions will use. Any suggestions on how to get this to work would be very much appreciated. Hi Daniel, 3 different contexts in your dial plan would work. Assign each block of accounts (rather than extensions) to the context with the routes that they should use. To change an account from using one trunk to another it would be as simple as changing its context. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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
[asterisk-users] IAX Calls - One Way Audio
Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for the whole organization, answering calls that come in for both locations. We have a problem where some calls (seemingly randomly) appear to get one way audio. This only happens for inbound calls off the PSTN, if they follow this pattern (which is a fair number of calls): Call comes in from PSTN to site A, gets put into a queue to be answered by receptionist as site B. Receptionist answers the call, and then puts the call on hold to perform an attended transfer to an extension at site A. (The call from the receptionist to the extension is OK). When the receptionist hits the 'transfer' button to actually transfer the call, the original caller cannot hear anything. The internal extension can hear the caller OK. This problem does not occur on every call. Since the issue has risen its head, I have enabled core, sip and iax debugging, but I am of yet unable to get the issue to occur on its own, to have a good look at the log files. FYI, I have disabled the asterisk Dial Commands in FreePBX, to solve another issue (where call audio bounces between the servers for a call that is transferred between sites and back again). Both servers are asterisk version 1.2.23, freepbx version 2.3.1.0. I have posted the contents of the iax.conf file below (which is identical on both servers). If there is any further information I can provide, please let me know and I can get this information. [general] disallow=all allow=g729 mailboxdetail=yes jitterbuffer=no ;maxjitterbuffer=500 ;jittershrinkrate=1 bandwidth=low tos=lowdelay trunk=yes notransfer=yes #include iax_general_custom.conf #include iax_registrations_custom.conf #include iax_registrations.conf #include iax_custom.conf #include iax_additional.conf Any suggestions are very welcome. Regards, Daniel ___ -- 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] IAX Calls - One Way Audio
Thanks Paul and Lyle for the suggestions. I would like to keep the phones configuration to one line for now, and see if I can solve the problem rather then just work around it. I have changed he notransfer option, will see what happens over the next few days. Thanks again for the suggestions, any further input is very much welcome. Many Thanks, Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Tuesday, 29 January 2008 11:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Calls - One Way Audio Does turning off the notransfer help? I would imagine that dropping the second server out of the equation might be useful, and save some bandwidth. PaulH On Tue, 2008-01-29 at 10:38 +1100, Daniel Cole wrote: Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for the whole organization, answering calls that come in for both locations. We have a problem where some calls (seemingly randomly) appear to get one way audio. This only happens for inbound calls off the PSTN, if they follow this pattern (which is a fair number of calls): Call comes in from PSTN to site A, gets put into a queue to be answered by receptionist as site B. Receptionist answers the call, and then puts the call on hold to perform an attended transfer to an extension at site A. (The call from the receptionist to the extension is OK). When the receptionist hits the 'transfer' button to actually transfer the call, the original caller cannot hear anything. The internal extension can hear the caller OK. This problem does not occur on every call. Since the issue has risen its head, I have enabled core, sip and iax debugging, but I am of yet unable to get the issue to occur on its own, to have a good look at the log files. FYI, I have disabled the asterisk Dial Commands in FreePBX, to solve another issue (where call audio bounces between the servers for a call that is transferred between sites and back again). Both servers are asterisk version 1.2.23, freepbx version 2.3.1.0. I have posted the contents of the iax.conf file below (which is identical on both servers). If there is any further information I can provide, please let me know and I can get this information. [general] disallow=all allow=g729 mailboxdetail=yes jitterbuffer=no ;maxjitterbuffer=500 ;jittershrinkrate=1 bandwidth=low tos=lowdelay trunk=yes notransfer=yes #include iax_general_custom.conf #include iax_registrations_custom.conf #include iax_registrations.conf #include iax_custom.conf #include iax_additional.conf Any suggestions are very welcome. Regards, Daniel ___ -- 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
Re: [asterisk-users] Asterisk scalability
Sorry to be a little OT.. But may I ask what some more of the specs are for that machine? Just trying to get an idea of what different hardware can achieve. Thanks, Daniel From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Thursday, 24 January 2008 7:57 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk scalability I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Asterisk is highly multi-threaded and definitely takes advantage of multiple cores. There are a few places where concurrency could be further improved, but its really quite good in 1.4. (IAX in 1.4 does handle traffic using a thread pool so will take advantage of multiple cores). By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. 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] Linksys SPA-9xx Audio Issues
We also use the Linksys SPA IP phones for our clients. We always change this setting to 0.020, which vastly improves audio performance. What are peoples thoughts on changing it to something lower, e.g. 0.010? Thanks, Daniel Cole -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Wednesday, 9 January 2008 9:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues On Tue, 2008-01-08 at 17:26 -0500, Andrew Joakimsen wrote: Anyone else have problems with phones like SPA-922, SPA-921, etc? If I remember correctly, the SPA-9XX phones default to sending packets every 30ms intead of every 20ms. Log in as Admin, click on the Advanced link, and go to the SIP tab. You'll find a setting labeled RTP Packet Size. Change it from 0.030 to 0.020 and see if that makes your audio quality better. It's done wonders for me in the past. -- Jared Smith Community Relations 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 ___ -- 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] Linksys SPA-9xx Audio Issues
Can you describe the issue more please? Can the remote person not hear you at all? Or is there distorted/broken voice? Cheers, Daniel Cole -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Wednesday, 9 January 2008 9:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Linksys SPA-9xx Audio Issues Anyone else have problems with phones like SPA-922, SPA-921, etc? Inbound audio is perfect but the remote end reports audio quality issues on the audio the handset is sending out. It's not the network I've tried asterisk 1.2, 1.4. I've used ulaw, G726, G793 G729. Ulaw seems to be the least problematic but its still an issue. Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI. I don't know it if happens all the time but about 40% of the time the remote caller reports they cannot hear me. ___ -- 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] Linksys SPA-9xx Audio Issues
I have found with a number of clients to who we have installed the LinkSys phones, that when you get the input gains to 6, that the phones have a tendency to pick up too much background noise. Have you experienced this at all? Cheers, Daniel Cole -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S Sent: Wednesday, 9 January 2008 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues The issues i have been having are probably similar to the original message, I use the Linksys 9XX Series phones and we used to always receive complaints from the person we were calling that they could hardly hear us. I fixed this by: Going into the Phone section of the config and setting the Handset, Speakerphone and Headset input gain to 6. And i also went into SIP and changed the RTP Packet Size to 0.020 This resolved the low volume issue, Sorry if you have a no sound issue, but thats how i resolved very low volume. Phones sound great now! Regards, Kevin Sandalin Daniel Cole wrote: Can you describe the issue more please? Can the remote person not hear you at all? Or is there distorted/broken voice? Cheers, Daniel Cole -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Wednesday, 9 January 2008 9:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Linksys SPA-9xx Audio Issues Anyone else have problems with phones like SPA-922, SPA-921, etc? Inbound audio is perfect but the remote end reports audio quality issues on the audio the handset is sending out. It's not the network I've tried asterisk 1.2, 1.4. I've used ulaw, G726, G793 G729. Ulaw seems to be the least problematic but its still an issue. Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI. I don't know it if happens all the time but about 40% of the time the remote caller reports they cannot hear me. ___ -- 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 -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ -- 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] Linksys SPA-9xx Audio Issues
Ok, no worries :) Most of our clients have a relatively open common work area, where the phones are located. I would be interested to know what your sales manager has experienced. Cheers, Daniel Cole -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S Sent: Wednesday, 9 January 2008 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues No, I haven't experienced this. I think were lucky because most voip phones are in there own offices, I will check with our sales manager this afternoon who sits in the call center and see what the background noise is like on her phone. I guess i'm just lucky that its a quiet environment, But there are a few people who *may* be affected and i will check this out and let you know. Regards, Kevin Daniel Cole wrote: I have found with a number of clients to who we have installed the LinkSys phones, that when you get the input gains to 6, that the phones have a tendency to pick up too much background noise. Have you experienced this at all? Cheers, Daniel Cole -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S Sent: Wednesday, 9 January 2008 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues The issues i have been having are probably similar to the original message, I use the Linksys 9XX Series phones and we used to always receive complaints from the person we were calling that they could hardly hear us. I fixed this by: Going into the Phone section of the config and setting the Handset, Speakerphone and Headset input gain to 6. And i also went into SIP and changed the RTP Packet Size to 0.020 This resolved the low volume issue, Sorry if you have a no sound issue, but thats how i resolved very low volume. Phones sound great now! Regards, Kevin Sandalin Daniel Cole wrote: Can you describe the issue more please? Can the remote person not hear you at all? Or is there distorted/broken voice? Cheers, Daniel Cole -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Wednesday, 9 January 2008 9:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Linksys SPA-9xx Audio Issues Anyone else have problems with phones like SPA-922, SPA-921, etc? Inbound audio is perfect but the remote end reports audio quality issues on the audio the handset is sending out. It's not the network I've tried asterisk 1.2, 1.4. I've used ulaw, G726, G793 G729. Ulaw seems to be the least problematic but its still an issue. Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI. I don't know it if happens all the time but about 40% of the time the remote caller reports they cannot hear me. ___ -- 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 -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ -- 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 -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ -- 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] Change Default Voicemail Message
Thank you for your reply Trevor. Is there an easy way to achieve this with a computer generated voice? We do not wish to manually record the messages if possible, in the interests of a consistent message across all voicemail boxes. What would be the easiest way to do this? Also, can you please give me some pointers on how to get the voicemail to play the separate message before the normal voicemail message? I'm guessing it would be done with a custom voicemail content, but im not sure how to write it correctly. Many Thanks, Daniel Cole From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor G. Hammonds Sent: Monday, 7 January 2008 3:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Change Default Voicemail Message Daniel, You could have Alison record a prompt Welcome to (nursing home) and re-record the prompt The person at extension... to be The person in room Then have your dialplan play the Welcome To... message before sending the call to voice mail. Then callers will hear Welcome to (Nursing Home). The person in room 5 is unavailable. Please leave your message... and if the resident has a recorded personal greeting or name, it would replace the The person... portion with either the resident's recorded name or greeting. Sincerely, Trevor Hammonds From: Daniel Cole Sent: Sunday, January 06, 2008 6:15 PM Hello List, I have a client (a nursing home) that we are looking at installing a trixbox for. One of the features that they would really like is a customized, standard voicemail recording for each of the residents rooms. We are looking for something along the lines of a voicemail recording like this: Welcome to (nursing home). You have reached room 5. Please leave a message after the tone. What would be the easiest way to get this to work. I have had a look at a few options, but I cant seem to find what I am after. Any help would be much appreciated. Thank You, Daniel Cole ___ --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] Change Default Voicemail Message
Hello List, I have a client (a nursing home) that we are looking at installing a trixbox for. One of the features that they would really like is a customized, standard voicemail recording for each of the residents rooms. We are looking for something along the lines of a voicemail recording like this: Welcome to (nursing home). You have reached room 5. Please leave a message after the tone. What would be the easiest way to get this to work. I have had a look at a few options, but I cant seem to find what I am after. Any help would be much appreciated. Thank You, Daniel Cole ___ --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] Dead Incoming call - Sangoma A200
Hello List, I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up. A bit of background: The client actually has two systems install (one at each location). Both systems are identical: Intel Server x3250 Dual Core Xenon Processor 1GB RAM Raid 1 160GB HDD's Sangoma A200 to interface to 2 PSTN lines. The odd thing is that only one of the systems is having this issue Making this one even more frustrating is that it only happens two or three times a day on this production box. Testing that I have done out of hours has been unable to re-produce the issue as of yet. I have enabled verbose debugging, and I have included below what I believe is the relevant section of the logs for when the issue occurs. As a quick overview, this was an incoming call on Zaptel channel 2, which was went to a ring group, and started to call SIP extension 203. This is when the call hung up. If there is any other information that I can provide to help, please let me know. This is a very frustrating issue... Log Output: Dec 21 13:43:48 NOTICE[15840] chan_zap.c: Got event 18 (Ring Begin)... Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing NoOp(Zap/2-1, Entering from-zaptel with DID == ) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Ringing(Zap/2-1, ) in new stack Dec 21 13:43:48 DEBUG[15840] chan_zap.c: Requested indication 3 on channel Zap/2-1 Dec 21 13:43:48 DEBUG[15840] pbx.c: Expression result is '1' Dec 21 13:43:48 DEBUG[15840] pbx.c: Function result is 's' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, DID=s) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing NoOp(Zap/2-1, DID is now s) in new stack Dec 21 13:43:48 DEBUG[15840] pbx.c: Expression result is '1' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing GotoIf(Zap/2-1, 1?zapok:notzap) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Goto (from-zaptel,s,8) Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing NoOp(Zap/2-1, Is a Zaptel Channel) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, CHAN=2-1) in new stack Dec 21 13:43:48 DEBUG[15840] pbx.c: Function result is '2' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, CHAN=2) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Macro(Zap/2-1, from-zaptel-2|s|1) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing NoOp(Zap/2-1, Entering macro-from-zaptel-2 with DID = s) in new stack Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: Noop Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Gosub(Zap/2-1, app-blacklist-check|s|1) in new stack Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: Gosub Dec 21 13:43:48 DEBUG[15840] app_macro.c: Incrementing gosub_level Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing LookupBlacklist(Zap/2-1, ) in new stack Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: LookupBlacklist Dec 21 13:43:48 DEBUG[15840] pbx.c: Expression result is '0' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing GotoIf(Zap/2-1, 0?blacklisted) in new stack Dec 21 13:43:48 DEBUG[15840] pbx.c: Not taking any branch Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: GotoIf Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Return(Zap/2-1, ) in new stack Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: Return Dec 21 13:43:48 DEBUG[15840] app_macro.c: Decrementing gosub_level Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Gosub(Zap/2-1, cidlookup|cidlookup_1|1) in new stack Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: Gosub Dec 21 13:43:48 DEBUG[15840] app_macro.c: Incrementing gosub_level Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing LookupCIDName(Zap/2-1, ) in new stack Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: LookupCIDName Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Return(Zap/2-1, ) in new stack Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: Return Dec 21 13:43:48 DEBUG[15840] app_macro.c: Decrementing gosub_level Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, __FROM_DID=s) in new stack Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: Set Dec 21 13:43:48 DEBUG[15840] pbx.c: Function result is '' Dec 21 13:43:48 DEBUG[15840] pbx.c: Expression result is '0' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing GotoIf(Zap/2-1, 0 ?cidok) in new stack Dec 21 13:43:48 DEBUG[15840] pbx.c: Not taking any branch Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: GotoIf Dec 21 13:43:48
Re: [asterisk-users] ip phone suggestion for Asia?
We currently also use the Linksys SPA942 and SPA963 IP phones. They are very nice phones, and very easy to manage. Cheers, Daniel Cole (CCNA) Technical Support Ph: 1800 424 683 Fax: 03 5221 7659 e: [EMAIL PROTECTED] w: hugonet.com.au --- The information transmitted is the property of HugoNet and is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the company. Any review, retransmission, dissemination and other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. P Please consider the environment before you print this e-mail or any attachments. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mindaugas Kezys Sent: Friday, 21 December 2007 6:03 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ip phone suggestion for Asia? Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive. We use them (SPA942) in our company. Everybody's happy. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of d tbsky Sent: Thursday, December 20, 2007 6:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ip phone suggestion for Asia? Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --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
Re: [asterisk-users] ip phone suggestion for Asia?
Hi, We currently use the 942 and 962. These both have backlight displays. The 962 is obviously the model with the nice color screen. These phones are very easy to manage. We configure all phone remotely via TFTP, and they phones are set to pull their config periodically. Very easy for making changes off-site :) Cheers, Daniel Cole (CCNA) Technical Support Ph: 1800 424 683 Fax: 03 5221 7659 e: [EMAIL PROTECTED] w: hugonet.com.au --- The information transmitted is the property of HugoNet and is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the company. Any review, retransmission, dissemination and other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. P Please consider the environment before you print this e-mail or any attachments. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Friday, 21 December 2007 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ip phone suggestion for Asia? Daniel Cole wrote: We currently also use the Linksys SPA942 and SPA963 IP phones. They are very nice phones, and very easy to manage. Both the 842 and 942 do not have lit displays and are hard to read. Is the 963 any better? Why anyone EVER built a LCD without backlighting is beyond me. John Novack Cheers, Daniel Cole (CCNA) Technical Support Ph: 1800 424 683 Fax: 03 5221 7659 e: [EMAIL PROTECTED] w: hugonet.com.au -- - The information transmitted is the property of HugoNet and is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the company. Any review, retransmission, dissemination and other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. P Please consider the environment before you print this e-mail or any attachments. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mindaugas Kezys Sent: Friday, 21 December 2007 6:03 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ip phone suggestion for Asia? Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive. We use them (SPA942) in our company. Everybody's happy. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of d tbsky Sent: Thursday, December 20, 2007 6:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ip phone suggestion for Asia? Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --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 -- Dog is my co-pilot ___ --Bandwidth and Colocation
Re: [asterisk-users] noun-verb vs verb-noun aka dogs black vs black dogs
My $0.02: This definitely sounds the way to do it. I agree that the order that the words are needed is almost irrelevant, as long as the whole lot is consistent. Good example too. Cheers Daniel Cole (CCNA) Technical Support Ph: 1800 424 683 Fax: 03 5221 7659 e: [EMAIL PROTECTED] w: hugonet.com.au --- The information transmitted is the property of HugoNet and is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the company. Any review, retransmission, dissemination and other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. P Please consider the environment before you print this e-mail or any attachments. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Millican Sent: Thursday, 20 December 2007 10:46 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] noun-verb vs verb-noun aka dogs black vs black dogs On Wednesday December 19 2007 6:09 pm, Tzafrir Cohen wrote: On Wed, Dec 19, 2007 at 02:47:39PM -0800, Steve Edwards wrote: This only works because you are closed to the alternative. The alternative (verb-noun) works fine for the above referenced applications and many more. Do you want to tally the number of users of applications that use noun-verb instead of verb-noun? Is there a reason verb-noun works fine for them and not for us? OK, here's a small usability test to your idea: Here's a partial list of actions from asterisk 1.4. Which of them is supported by your hypothetical MGCP device? (no cheating, please) active add answer audit autoanswer boost clear convert del deltree dial dumphtml flash get hangup logoff mute put reload remove save send set show showkey transfer unmute Okay I have to put my 2 cents in now can't resist any longer even though it may only be worth 0.5 cents. In MY opinion, consistency is first and formost. I can learn almost any command struture IF i put my mind to it and I want to do so. What is hard for me is changing in mid stream. having said that I always liked a drill down structure. Big idea first, followed by category of idea, followed by.. and so on till you get the the exact single item that you are looking for. A US based example: show world north_america us state nh capitol Gives: Concord You could easily do : show world giving all the continents show world north_america giving all countries in North America and so on down the line. To ME and maybe only me, this make since, object world knows of continents, object continents knows of countries, object countries knows of state, object state knows of capitols. Easy for programmers, users and computers alike. again just my opinion. JohnM ___ --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
[asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue?
Hello Everyone, We have recently installed a pair of Trixbox servers in for a client of our. They have two locations, with one server each. The servers terminate 3 standard POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 all around. Each site has two (2) 512k/512k ADSL connections terminating into a Cisco 877W router (using an additional 'dumb' modem in a separate VLAN for the extra dsl connection). Using policy based routing, all Voice Data goes over one DSL connection (the one that terminates directly into the router), and all other traffic (e.g. Web and VPN) goes out the second connection (the bridged dumb dsl modem). We are also the ISP for this client, and as thus we have full monitoring of our Layer 2 and Layer 3 networks. From our analysis, it doesn't appear that there is any issue in these networks. We have other customers using the VoIP service, who have not complained of these issues. Now for the Fun part! The client is complaining of issues with inter-site calls. They are reporting issues with crackly and broken speech, and horrible jitter (or packet loss). This presents a huge issues, because they have one receptionist answering all calls for both sites. So if a call comes in from the other site, it automatically an inter-site call, and the quality falls out of it. If the call is then transfered back to the originating site, the audio 'bounces' between the two sites, which add to the call quality degradation. We have been monitoring the router while these incidents have been reported, and it does not appear to be a bandwidth issue. The DSL tail used for Voice gets to no more then 120k in each direction (we have tested the links, and can pull data at 53k/s between sites). CPU usage floats at around 20-25% under load. The router has only shows major packet loss (that we can tell) when REALLY pushing it in testing (e.g. 10+ calls between sites). We have enabled the SIP jitter buffer, as well as the IAX jitter buffer, which appeared to make a huge difference, but the issue is still ongoing. These issues have also been reported with some outbound VoIP calls. Internal calls, and calls directly in or out of the Sangoma card are clear, with no issues reported. Does anyone have any thoughts on what could be causing these issues? We have been racking our brains here, and have tried everything that we can think of. These system is a million times better then what is what when it was first installed, but it is still not where it should be in terms of quality. Any thoughts/ideas are most welcome. Thank you Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. ___ --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] Call Quality Issues With 2 Trixbox's - RouterIssue?
The two boxes are labeled as per the town they are in: Leongatha and Korumburra. The receptionist is in Korumburra. When a call comes in off the PSTN in Leongatha, the first number in the call queue is the receptionist. If she answers it, then the media flow looks like this: PSTN - Leongatha - IAX Trunk - Korumburra - Receptionist Phone If she then transfers the call back to a Leongatha extension, the media path looks like this: PSTN - Leongatha - IAX Trunk - Korumburra - IAX Trunk - Leongatha - IP Phone I believe that it is possible to stop this 'bouncing' of the call from happening by using the re-invite feature. However, taking the Trixbox's out of the media path is undesirable, as they client needs to be able to record calls. Also, doing this does not 'fix' the underlying problem. They are also having some issues with outbound calls from Leongatha (over VoIP), and they are having no real issues at Korumburra. Many Thanks, Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, 12 December 2007 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - RouterIssue? How are the calls being transferred from Box A to Box B? On what box is the receptionist registered too? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Cole Sent: Tuesday, December 11, 2007 9:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call Quality Issues With 2 Trixbox's - RouterIssue? Hello Everyone, We have recently installed a pair of Trixbox servers in for a client of our. They have two locations, with one server each. The servers terminate 3 standard POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 all around. Each site has two (2) 512k/512k ADSL connections terminating into a Cisco 877W router (using an additional 'dumb' modem in a separate VLAN for the extra dsl connection). Using policy based routing, all Voice Data goes over one DSL connection (the one that terminates directly into the router), and all other traffic (e.g. Web and VPN) goes out the second connection (the bridged dumb dsl modem). We are also the ISP for this client, and as thus we have full monitoring of our Layer 2 and Layer 3 networks. From our analysis, it doesn't appear that there is any issue in these networks. We have other customers using the VoIP service, who have not complained of these issues. Now for the Fun part! The client is complaining of issues with inter-site calls. They are reporting issues with crackly and broken speech, and horrible jitter (or packet loss). This presents a huge issues, because they have one receptionist answering all calls for both sites. So if a call comes in from the other site, it automatically an inter-site call, and the quality falls out of it. If the call is then transfered back to the originating site, the audio 'bounces' between the two sites, which add to the call quality degradation. We have been monitoring the router while these incidents have been reported, and it does not appear to be a bandwidth issue. The DSL tail used for Voice gets to no more then 120k in each direction (we have tested the links, and can pull data at 53k/s between sites). CPU usage floats at around 20-25% under load. The router has only shows major packet loss (that we can tell) when REALLY pushing it in testing (e.g. 10+ calls between sites). We have enabled the SIP jitter buffer, as well as the IAX jitter buffer, which appeared to make a huge difference, but the issue is still ongoing. These issues have also been reported with some outbound VoIP calls. Internal calls, and calls directly in or out of the Sangoma card are clear, with no issues reported. Does anyone have any thoughts on what could be causing these issues? We have been racking our brains here, and have tried everything that we can think of. These system is a million times better then what is what when it was first installed, but it is still not where it should be in terms of quality. Any thoughts/ideas are most welcome. Thank you Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. ___ --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] Call Quality Issues With 2 Trixbox's - Router Issue?
G729 All Around. Daniel Cole (CCNA) Technical Support [http://www.hugonet.com.au/clients/hugonet.gif] Ph: 1800 424 683 Fax: 03 5221 7659 e: [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] w: hugonet.com.auhttp://www.hugonet.com.au/ --- The information transmitted is the property of HugoNet and is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the company. Any review, retransmission, dissemination and other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. P Please consider the environment before you print this e-mail or any attachments. From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Wednesday, 12 December 2007 4:10 PM To: Daniel Cole Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue? What codec are you using? PaulH On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote: Hello Everyone, We have recently installed a pair of Trixbox servers in for a client of our. They have two locations, with one server each. The servers terminate 3 standard POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 all around. Each site has two (2) 512k/512k ADSL connections terminating into a Cisco 877W router (using an additional 'dumb' modem in a separate VLAN for the extra dsl connection). Using policy based routing, all Voice Data goes over one DSL connection (the one that terminates directly into the router), and all other traffic (e.g. Web and VPN) goes out the second connection (the bridged dumb dsl modem). We are also the ISP for this client, and as thus we have full monitoring of our Layer 2 and Layer 3 networks. From our analysis, it doesn't appear that there is any issue in these networks. We have other customers using the VoIP service, who have not complained of these issues. Now for the Fun part! The client is complaining of issues with inter-site calls. They are reporting issues with crackly and broken speech, and horrible jitter (or packet loss). This presents a huge issues, because they have one receptionist answering all calls for both sites. So if a call comes in from the other site, it automatically an inter-site call, and the quality falls out of it. If the call is then transfered back to the originating site, the audio 'bounces' between the two sites, which add to the call quality degradation. We have been monitoring the router while these incidents have been reported, and it does not appear to be a bandwidth issue. The DSL tail used for Voice gets to no more then 120k in each direction (we have tested the links, and can pull data at 53k/s between sites). CPU usage floats at around 20-25% under load. The router has only shows major packet loss (that we can tell) when REALLY pushing it in testing (e.g. 10+ calls between sites). We have enabled the SIP jitter buffer, as well as the IAX jitter buffer, which appeared to make a huge difference, but the issue is still ongoing. These issues have also been reported with some outbound VoIP calls. Internal calls, and calls directly in or out of the Sangoma card are clear, with no issues reported. Does anyone have any thoughts on what could be causing these issues? We have been racking our brains here, and have tried everything that we can think of. These system is a million times better then what is what when it was first installed, but it is still not where it should be in terms of quality. Any thoughts/ideas are most welcome. Thank you Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. ___ --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 inline: hugonet.gif___ --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] Call Quality Issues With 2 Trixbox's - Router Issue?
Hi Paul, Where abouts exactly is the best place to get these figures from? I have been checking iax2 show netstats, which does give some figures. These appear not to be accurate though, as when there are multiple inter-site calls, the result for one channel of audio can show no jitter or latency, but another will have some jitter and latency. Or is this a weird way for the problem to show its head? Thanks, Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Wednesday, 12 December 2007 4:40 PM To: Daniel Cole Subject: RE: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue? Hmmm..wierd Are you getting an weird jitter/latency figures in the CLI? PaulH On Wed, 2007-12-12 at 16:37 +1100, Daniel Cole wrote: G729 All Around. Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Wednesday, 12 December 2007 4:10 PM To: Daniel Cole Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue? What codec are you using? PaulH On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote: Hello Everyone, We have recently installed a pair of Trixbox servers in for a client of our. They have two locations, with one server each. The servers terminate 3 standard POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 all around. Each site has two (2) 512k/512k ADSL connections terminating into a Cisco 877W router (using an additional 'dumb' modem in a separate VLAN for the extra dsl connection). Using policy based routing, all Voice Data goes over one DSL connection (the one that terminates directly into the router), and all other traffic (e.g. Web and VPN) goes out the second connection (the bridged dumb dsl modem). We are also the ISP for this client, and as thus we have full monitoring of our Layer 2 and Layer 3 networks. From our analysis, it doesn't appear that there is any issue in these networks. We have other customers using the VoIP service, who have not complained of these issues. Now for the Fun part! The client is complaining of issues with inter-site calls. They are reporting issues with crackly and broken speech, and horrible jitter (or packet loss). This presents a huge issues, because they have one receptionist answering all calls for both sites. So if a call comes in from the other site, it automatically an inter-site call, and the quality falls out of it. If the call is then transfered back to the originating site, the audio 'bounces' between the two sites, which add to the call quality degradation. We have been monitoring the router while these incidents have been reported, and it does not appear to be a bandwidth issue. The DSL tail used for Voice gets to no more then 120k in each direction (we have tested the links, and can pull data at 53k/s between sites). CPU usage floats at around 20-25% under load. The router has only shows major packet loss (that we can tell) when REALLY pushing it in testing (e.g. 10+ calls between sites). We have enabled the SIP jitter buffer, as well as the IAX jitter buffer, which appeared to make a huge difference, but the issue is still ongoing. These issues have also been reported with some outbound VoIP calls. Internal calls, and calls directly in or out of the Sangoma card are clear, with no issues reported. Does anyone have any thoughts on what could be causing these issues? We have been racking our brains here, and have tried everything that we can think of. These system is a million times better then what is what when it was first installed, but it is still not where it should be in terms of quality. Any thoughts/ideas are most welcome. Thank you Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. ___ --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
