Re: [asterisk-users] local channels and g729a voice quality
On 16/01/12 07:59, Roi Stork wrote: I also asked my provider to test call me using their Cisco as5300 system and g729 codec and compared it with ulaw. The difference is unnoticable. ^^ this doesn't make any sense, the difference *should* be very much noticeable. g729 is a lower quality codec (in terms of audio quality). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels and g729a voice quality
On 01/16/2012 03:59 PM, Roi Stork wrote: Hi, We noticed a very sharp drop in voice quality when using digium g729a codec. The problem seems to happen if the A channel (caller's channel) is a landline/mobile number contacted using the same outgoing provider (as a local channel). It sounds like listening to a mono speaker on low volume. You have conflated two very different things there - landline calls and cellular calls. A land line to a VoIP user by G.729A should sounds pretty good. A cellphone to a VoIP user by G.729A should sound *far* worse. Converting between two different low bit rate codecs really hits the quality, and all cellphone calls are low bit rate. If I use a softphone that is directly registered to our asterisk box the audio quality improves, the words come out more clearer and louder. You are conflating two things again. Quality and volume are largely independent issues. I also asked my provider to test call me using their Cisco as5300 system and g729 codec and compared it with ulaw. The difference is unnoticable. Try that again. If you really can't hear the difference you should check carefully that the system is working as you think it is. If it is, maybe you should consult a doctor. G.729A is considerably poorer than ulaw. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels and g729a voice quality
Hi, I wasn't aware of the difference in quality between landline and mobile phones, or that cellphones use a low bit rate. I did a test again, landline voice quality is better. From what you observed, how much drop in quality do I expect when switching from ulaw to g729 for a normal mobile-to-mobile call? On Tue, Jan 17, 2012 at 9:37 PM, Steve Underwood ste...@coppice.org wrote: On 01/16/2012 03:59 PM, Roi Stork wrote: Hi, We noticed a very sharp drop in voice quality when using digium g729a codec. The problem seems to happen if the A channel (caller's channel) is a landline/mobile number contacted using the same outgoing provider (as a local channel). It sounds like listening to a mono speaker on low volume. You have conflated two very different things there - landline calls and cellular calls. A land line to a VoIP user by G.729A should sounds pretty good. A cellphone to a VoIP user by G.729A should sound *far* worse. Converting between two different low bit rate codecs really hits the quality, and all cellphone calls are low bit rate. If I use a softphone that is directly registered to our asterisk box the audio quality improves, the words come out more clearer and louder. You are conflating two things again. Quality and volume are largely independent issues. I also asked my provider to test call me using their Cisco as5300 system and g729 codec and compared it with ulaw. The difference is unnoticable. Try that again. If you really can't hear the difference you should check carefully that the system is working as you think it is. If it is, maybe you should consult a doctor. G.729A is considerably poorer than ulaw. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels and g729a voice quality
On 18-01-12 04:57, Roi Stork wrote: Hi, I wasn't aware of the difference in quality between landline and mobile phones, or that cellphones use a low bit rate. I did a test again, landline voice quality is better. From what you observed, how much drop in quality do I expect when switching from ulaw to g729 for a normal mobile-to-mobile call? Afaik mobile to mobile uses the gsm codec (in gsm country, no idea about cdma etc.). IMHO gsm is not bad but certainly not as good as ulaw/alaw. Read up on MOS: https://en.wikipedia.org/wiki/Mean_opinion_score Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] local channels and g729a voice quality
Hi, We noticed a very sharp drop in voice quality when using digium g729a codec. The problem seems to happen if the A channel (caller's channel) is a landline/mobile number contacted using the same outgoing provider (as a local channel). It sounds like listening to a mono speaker on low volume. If I use a softphone that is directly registered to our asterisk box the audio quality improves, the words come out more clearer and louder. I also asked my provider to test call me using their Cisco as5300 system and g729 codec and compared it with ulaw. The difference is unnoticable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] local channels
I am using the AMI to dispatch (2) calls to Local/my_prior...@my_context where: [my_context] exten = my_priority,1,Answer() exten = my_priority,n,Dial(${LOCAL_DIAL}) and LOCAL_DIAL has the actual phone number to dial. The first call goes through just fine and I see DAHDI/1/ being called. The second call I see DAHDI/2/ and a message about everyone is busy on congested. I presume I can have more than one local channel active? My AMI channel line is: Channel: Local/my_prior...@my_context for both calls. I have a Variable with the LOCAL_DIAL set. I am using DAHDI 2.2.0 with libpri 1.4.10.2 and asterisk 1.4.26.2 With a PRI connection. All normal calls to phones work fine. When I make my (2) local calls all 23 lines are idle. Is there something I am missing? Why would I not be able to make 2 local channel calls at the same time? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels
Jerry Geis wrote: I am using the AMI to dispatch (2) calls to Local/my_prior...@my_context where: [my_context] exten = my_priority,1,Answer() exten = my_priority,n,Dial(${LOCAL_DIAL}) and LOCAL_DIAL has the actual phone number to dial. The first call goes through just fine and I see DAHDI/1/ being called. The second call I see DAHDI/2/ and a message about everyone is busy on congested. I presume I can have more than one local channel active? My AMI channel line is: Channel: Local/my_prior...@my_context for both calls. I have a Variable with the LOCAL_DIAL set. That is correct. It sounds like your need to make sure you're using the same trunk group within DAHDI over and over: Dial(DAHDI/1/${LOCAL_DIAL}) -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels
Jerry Geis wrote: / I am using the AMI to dispatch (2) calls to Local/my_priority at my_context http://lists.digium.com/mailman/listinfo/asterisk-users // where: // [my_context] // exten = my_priority,1,Answer() // exten = my_priority,n,Dial(${LOCAL_DIAL}) // // and LOCAL_DIAL has the actual phone number to dial. // // The first call goes through just fine and I see DAHDI/1/ being // called. The second call I see // DAHDI/2/ and a message about everyone is busy on congested. // // I presume I can have more than one local channel active? My AMI channel // line is: // Channel: Local/my_priority at my_context http://lists.digium.com/mailman/listinfo/asterisk-users for both calls. I have a Variable // with the LOCAL_DIAL set. / That is correct. It sounds like your need to make sure you're using the same trunk group within DAHDI over and over: Dial(DAHDI/1/${LOCAL_DIAL}) Alex, My Dial() command is Dial($LOCAL_DIAL) and for the first call it is DAHDI/1/ and for the second call it is DAHDI/2/XXX. My LOCAL_DIAL has the complete dial command DAHDI/xx/ So I am using line 1 and line 2 of the PRI connection. I dont see why the second call is saying - everyone busy or congested at this time. These are the only 2 calls active. One on line 1 and one on line 2. Only 2 of the 23 lines available am I using. All 23 lines are in the same group. How do I tell why it thinks its busy? Thanks, Jerry ___ -- 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] local channels
I think the problem is that the way this works - if I'm not mistaken - is that the attribute after the first delimeter in the channel string is a trunk group and not a channel. In other words, DAHDI/1 refers to circuit 1, not B-channel 1 of circuit 1. B-channel 1 would be DAHDI/1/1. Jerry Geis wrote: Jerry Geis wrote: / I am using the AMI to dispatch (2) calls to Local/my_priority at my_context http://lists.digium.com/mailman/listinfo/asterisk-users // where: // [my_context] // exten = my_priority,1,Answer() // exten = my_priority,n,Dial(${LOCAL_DIAL}) // // and LOCAL_DIAL has the actual phone number to dial. // // The first call goes through just fine and I see DAHDI/1/ being // called. The second call I see // DAHDI/2/ and a message about everyone is busy on congested. // // I presume I can have more than one local channel active? My AMI channel // line is: // Channel: Local/my_priority at my_context http://lists.digium.com/mailman/listinfo/asterisk-users for both calls. I have a Variable // with the LOCAL_DIAL set. / That is correct. It sounds like your need to make sure you're using the same trunk group within DAHDI over and over: Dial(DAHDI/1/${LOCAL_DIAL}) Alex, My Dial() command is Dial($LOCAL_DIAL) and for the first call it is DAHDI/1/ and for the second call it is DAHDI/2/XXX. My LOCAL_DIAL has the complete dial command DAHDI/xx/ So I am using line 1 and line 2 of the PRI connection. I dont see why the second call is saying - everyone busy or congested at this time. These are the only 2 calls active. One on line 1 and one on line 2. Only 2 of the 23 lines available am I using. All 23 lines are in the same group. How do I tell why it thinks its busy? Thanks, Jerry -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels
My Dial() command is Dial($LOCAL_DIAL) Perhaps you should be using: Dial(${LOCAL_DIAL}) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels
/ My Dial() command is Dial($LOCAL_DIAL) / Perhaps you should be using: Dial(${LOCAL_DIAL}) Steve, Thanks I tried that also and same result. Jerry ___ -- 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] local channels
This is what I see: -- Executing [my_prior...@my_context:1] Answer(Local/my_prior...@my_context-90d5,2, ) in new stack -- Executing [my_prior...@my_context:2] Dial(Local/my_prior...@my_context-90d5,2, DAHDI/3/4000) in new stack [Nov 9 16:25:17] WARNING[8979]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Local/my_prior...@my_context-90d5,2' status is 'CHANUNAVAIL' Normal calls all work just fine. I can call into the box and out the box to extensions and cell phones. When I place this call all lines are idle. My call through AMI is basically this: Action: Originate Async: yes Channel: Local/my_prior...@my_context Context: my_context Application: AGI Variable: LOCAL_DIAL=DAHDI/4/4001 Data: smvoice Priority: 1 Any ideas why the all channels busy and unable to create channel? Jerry ___ -- 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] local channels
LOCAL_DIAL is populated - exten = s,1,Verbose(call ${LOCAL_DIAL}) - exten = s,2,Dial(${LOCAL_DIAL}) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, November 09, 2009 3:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] local channels / My Dial() command is Dial($LOCAL_DIAL) / Perhaps you should be using: Dial(${LOCAL_DIAL}) Steve, Thanks I tried that also and same result. Jerry ___ -- 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] local channels
So 4001 is a local FXS DAHDI channel? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, November 09, 2009 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] local channels This is what I see: -- Executing [my_prior...@my_context:1] Answer(Local/my_prior...@my_context-90d5,2, ) in new stack -- Executing [my_prior...@my_context:2] Dial(Local/my_prior...@my_context-90d5,2, DAHDI/3/4000) in new stack [Nov 9 16:25:17] WARNING[8979]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Local/my_prior...@my_context-90d5,2' status is 'CHANUNAVAIL' Normal calls all work just fine. I can call into the box and out the box to extensions and cell phones. When I place this call all lines are idle. My call through AMI is basically this: Action: Originate Async: yes Channel: Local/my_prior...@my_context Context: my_context Application: AGI Variable: LOCAL_DIAL=DAHDI/4/4001 Data: smvoice Priority: 1 Any ideas why the all channels busy and unable to create channel? Jerry ___ -- 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