[Asterisk-Users] iconnect quality?
Does anyone here use iconnect regularly with Asterisk? If so, what do you think of its reliability and quality? I used up my 10 free minutes just getting it to work. By the end it was working, but I found that (1) many calls did not connect, due to a variety of errors reported by them (service unavailable and such) and (2) when I did connect, I could be heard but I could not hear the person I was talking to. I would appreciate hearing any experiences. Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
On Mon, 2003-03-10 at 03:51, Jim Archer wrote: > Does anyone here use iconnect regularly with Asterisk? If so, what do you > think of its reliability and quality? I used up my 10 free minutes just > getting it to work. > > By the end it was working, but I found that (1) many calls did not connect, > due to a variety of errors reported by them (service unavailable and such) > and (2) when I did connect, I could be heard but I could not hear the > person I was talking to. > > I would appreciate hearing any experiences. I have had no problems with call quality, but right now inbound calling from them is having an issue with completing the call through to asterisk. Call quality wise though, no, I've been very happy with it. -- William Walsh <[EMAIL PROTECTED]> Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Iconnect uses codecs g723 and g711 that can be configured for each account (you can change them by the prefix) With their dialer and g723 I can here just fine (I have a 64k broadband connection). With their dialer and g711 the quality suffers greatly. With * and GSM I cannot here anything (and don´t know if they can here me). The call gets logged on iconnect´s CDR. Upon looking at the SIP debug everything appears just fine, but again, I cannot hear anything. With * and ulaw or mlaw I can hear the first 5 secs and nothing else. I have tried FWD (not going through *) which uses g711 and I have had nor problem whatsoever. - Original Message - From: "Jim Archer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 10, 2003 8:51 AM Subject: [Asterisk-Users] iconnect quality? > Does anyone here use iconnect regularly with Asterisk? If so, what do you > think of its reliability and quality? I used up my 10 free minutes just > getting it to work. > > By the end it was working, but I found that (1) many calls did not connect, > due to a variety of errors reported by them (service unavailable and such) > and (2) when I did connect, I could be heard but I could not hear the > person I was talking to. > > I would appreciate hearing any experiences. > > Thanks! > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
On Mon, 2003-03-10 at 13:47, Dan Fernandez wrote: > Iconnect uses codecs g723 and g711 that can be configured for each account > (you can change them by the prefix) > > With their dialer and g723 I can here just fine (I have a 64k broadband > connection). With their dialer and g711 the quality suffers greatly. > > With * and GSM I cannot here anything (and don´t know if they can here me). > The call gets logged on iconnect´s CDR. Upon looking at the SIP debug > everything appears just fine, but again, I cannot hear anything. g711 is 8 bit 8khz, or 64Kbit of audio data alone without the overhead of TCP/IP nor SIP. 64Kbit is not broadband, it is a DS0. It may be a tad over dial up, but please don't consider it broadband. This would explain your quality problem, you fill the link in the first set of samples, and the rest is queued up and therefore does not arrive like a stream should. Each fram probably adds a millisecond or more to the stream and soon enough you are so starved for audio data that it should give up. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Wow, this looks like a way to configure Asterisk to solve this not hearing the called person problem, but I have to confess, I can't figure out how to actually do what you suggest. I assume I need some entries in sip.conf? Sorry to ask for more detail like this, but I am now and don't know how to configure sip. Thanks... --On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez <[EMAIL PROTECTED]> wrote: Iconnect uses codecs g723 and g711 that can be configured for each account (you can change them by the prefix) With their dialer and g723 I can here just fine (I have a 64k broadband connection). With their dialer and g711 the quality suffers greatly. With * and GSM I cannot here anything (and don´t know if they can here me). The call gets logged on iconnect´s CDR. Upon looking at the SIP debug everything appears just fine, but again, I cannot hear anything. With * and ulaw or mlaw I can hear the first 5 secs and nothing else. I have tried FWD (not going through *) which uses g711 and I have had nor problem whatsoever. - Original Message - From: "Jim Archer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 10, 2003 8:51 AM Subject: [Asterisk-Users] iconnect quality? Does anyone here use iconnect regularly with Asterisk? If so, what do you think of its reliability and quality? I used up my 10 free minutes just getting it to work. By the end it was working, but I found that (1) many calls did not connect, due to a variety of errors reported by them (service unavailable and such) and (2) when I did connect, I could be heard but I could not hear the person I was talking to. I would appreciate hearing any experiences. Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Yes, 64K is not much of a broadband and believe it or not, I am paying US$60 for it (I believe this is the case in many 3rd world countries). Is there a codec translator between GSM and g723? How come I can use FWD just fine with g711? - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 10, 2003 5:33 PM Subject: Re: [Asterisk-Users] iconnect quality? > On Mon, 2003-03-10 at 13:47, Dan Fernandez wrote: > > Iconnect uses codecs g723 and g711 that can be configured for each account > > (you can change them by the prefix) > > > > With their dialer and g723 I can here just fine (I have a 64k broadband > > connection). With their dialer and g711 the quality suffers greatly. > > > > With * and GSM I cannot here anything (and don´t know if they can here me). > > The call gets logged on iconnect´s CDR. Upon looking at the SIP debug > > everything appears just fine, but again, I cannot hear anything. > > g711 is 8 bit 8khz, or 64Kbit of audio data alone without the overhead > of TCP/IP nor SIP. 64Kbit is not broadband, it is a DS0. It may be a tad > over dial up, but please don't consider it broadband. > > This would explain your quality problem, you fill the link in the first > set of samples, and the rest is queued up and therefore does not arrive > like a stream should. Each fram probably adds a millisecond or more to > the stream and soon enough you are so starved for audio data that it > should give up. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Dan, welcome to the club. Country like Poland which is supposed to get into EU is also 3rd world in this matter. (However things are changing a lot lately) Regards, Krzysztof Bujak - Original Message - From: "Dan Fernandez" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 10, 2003 11:18 PM Subject: Re: [Asterisk-Users] iconnect quality? > Yes, 64K is not much of a broadband and believe it or not, I am paying US$60 > for it (I believe this is the case in many 3rd world countries). > > Is there a codec translator between GSM and g723? > > How come I can use FWD just fine with g711? > > - Original Message - > From: "Steven Critchfield" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, March 10, 2003 5:33 PM > Subject: Re: [Asterisk-Users] iconnect quality? > > > > On Mon, 2003-03-10 at 13:47, Dan Fernandez wrote: > > > Iconnect uses codecs g723 and g711 that can be configured for each > account > > > (you can change them by the prefix) > > > > > > With their dialer and g723 I can here just fine (I have a 64k broadband > > > connection). With their dialer and g711 the quality suffers greatly. > > > > > > With * and GSM I cannot here anything (and don´t know if they can here > me). > > > The call gets logged on iconnect´s CDR. Upon looking at the SIP debug > > > everything appears just fine, but again, I cannot hear anything. > > > > g711 is 8 bit 8khz, or 64Kbit of audio data alone without the overhead > > of TCP/IP nor SIP. 64Kbit is not broadband, it is a DS0. It may be a tad > > over dial up, but please don't consider it broadband. > > > > This would explain your quality problem, you fill the link in the first > > set of samples, and the rest is queued up and therefore does not arrive > > like a stream should. Each fram probably adds a millisecond or more to > > the stream and soon enough you are so starved for audio data that it > > should give up. > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > === > Ta wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez system antywirusowy na serwerze IT Form. > === Ta wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez system antywirusowy na serwerze IT Form.
Re: [Asterisk-Users] iconnect quality?
Dan, I am having similar issues. I don't quite understand how you change the codec at iconnect (i.e. what do you mean by using the prefix?). Gregg On Mon, 2003-03-10 at 14:47, Dan Fernandez wrote: > Iconnect uses codecs g723 and g711 that can be configured for each account > (you can change them by the prefix) > > With their dialer and g723 I can here just fine (I have a 64k broadband > connection). With their dialer and g711 the quality suffers greatly. > > With * and GSM I cannot here anything (and don´t know if they can here me). > The call gets logged on iconnect´s CDR. Upon looking at the SIP debug > everything appears just fine, but again, I cannot hear anything. > > With * and ulaw or mlaw I can hear the first 5 secs and nothing else. > > I have tried FWD (not going through *) which uses g711 and I have had nor > problem whatsoever. > > - Original Message - > From: "Jim Archer" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, March 10, 2003 8:51 AM > Subject: [Asterisk-Users] iconnect quality? > > > > Does anyone here use iconnect regularly with Asterisk? If so, what do you > > think of its reliability and quality? I used up my 10 free minutes just > > getting it to work. > > > > By the end it was working, but I found that (1) many calls did not > connect, > > due to a variety of errors reported by them (service unavailable and such) > > and (2) when I did connect, I could be heard but I could not hear the > > person I was talking to. > > > > I would appreciate hearing any experiences. > > > > Thanks! > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Dan, I am having similar issues. I don't quite understand how you change the codec at iconnect (i.e. what do you mean by using the prefix?). Gregg On Mon, 2003-03-10 at 14:47, Dan Fernandez wrote: > Iconnect uses codecs g723 and g711 that can be configured for each account > (you can change them by the prefix) > > With their dialer and g723 I can here just fine (I have a 64k broadband > connection). With their dialer and g711 the quality suffers greatly. > > With * and GSM I cannot here anything (and don´t know if they can here me). > The call gets logged on iconnect´s CDR. Upon looking at the SIP debug > everything appears just fine, but again, I cannot hear anything. > > With * and ulaw or mlaw I can hear the first 5 secs and nothing else. > > I have tried FWD (not going through *) which uses g711 and I have had nor > problem whatsoever. > > - Original Message - > From: "Jim Archer" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, March 10, 2003 8:51 AM > Subject: [Asterisk-Users] iconnect quality? > > > > Does anyone here use iconnect regularly with Asterisk? If so, what do you > > think of its reliability and quality? I used up my 10 free minutes just > > getting it to work. > > > > By the end it was working, but I found that (1) many calls did not > connect, > > due to a variety of errors reported by them (service unavailable and such) > > and (2) when I did connect, I could be heard but I could not hear the > > person I was talking to. > > > > I would appreciate hearing any experiences. > > > > Thanks! > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
--On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez <[EMAIL PROTECTED]> wrote: Iconnect uses codecs g723 and g711 that can be configured for each account (you can change them by the prefix) I tried adding the in front of a number and it reliably generates error "488 invalid media." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
On Mon, 2003-03-10 at 05:13, William X Walsh wrote: > I have had no problems with call quality, but right now inbound calling > from them is having an issue with completing the call through to > asterisk. Just to make sure the other d3/iconnecthere users on the list know, this problem has been fixed in the CVS as of this morning. Thanks to Mark and Ravi for finding a way to make * work with d3's inbound calls and committing it to CVS so quickly. -- William Walsh <[EMAIL PROTECTED]> Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Jim, I changed my extensions entry for iconnect to: exten => _1XX,1,Dial,SIP/[EMAIL PROTECTED] and my calls work fine with ulaw. I am calling from a linejack card with format=ulaw and SIP with allow=ulaw. Gregg On Mon, 2003-03-10 at 23:01, Jim Archer wrote: > --On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez > <[EMAIL PROTECTED]> wrote: > > > Iconnect uses codecs g723 and g711 that can be configured for each account > > (you can change them by the prefix) > > I tried adding the in front of a number and it reliably generates > error "488 invalid media." > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Hi Greg and thanks very much... A few questions... First, regarding the prefix, it seemed that this acts as a toggle, switching from the one codec to the other. But how do I set which me system uses by default? Or does iconnect always use the high bandwidth one by default (such that the always switches to the low bandwidth one)? Next, I am still struggling to understand the SIP options and what goes where. Could you please tell me where the format command goes? Is this an option on the channel? I thing the allow goes in sip.conf. Finally, does this have any impact on the problem where the person called can not be heard? Thanks!!! Jim --On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz <[EMAIL PROTECTED]> wrote: Jim, I changed my extensions entry for iconnect to: exten => _1XX,1,Dial,SIP/[EMAIL PROTECTED] and my calls work fine with ulaw. I am calling from a linejack card with format=ulaw and SIP with allow=ulaw. Gregg On Mon, 2003-03-10 at 23:01, Jim Archer wrote: --On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez <[EMAIL PROTECTED]> wrote: > Iconnect uses codecs g723 and g711 that can be configured for each > account (you can change them by the prefix) I tried adding the in front of a number and it reliably generates error "488 invalid media." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
I haven't play around enough to know whether or not the prefix is a toggle. I will do some experimenting and let you know. Right now I am prefixing all my calls with . My experience is that when the carrier's format is G723.1, you can't hear the incoming voice. When it is in G711 you can. I have made several calls using G711 and they are acceptable quality. Note that if you disallow=gsm in the sip.conf file you will get the 488 media errors you reported earlier. Below are my config files for sip and the linejack cards: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context=iconnect; Default for incoming calls allow=gsm allow=ulaw allow=alaw ;register=1813342:[EMAIL PROTECTED] ;register=1202454:[EMAIL PROTECTED] [iconnecthere] type=friend username= secret=XXX host=sipauth.deltathree.com ; ; Linux Telephony Interface ; ; Configuration file ; [interfaces] mode=dialtone format=ulaw echocancel=medium silencesupression=no context=local context=default txgain=100% rxgain=100% device => /dev/phone0 On Tue, 2003-03-11 at 14:28, Jim Archer wrote: > Hi Greg and thanks very much... > > A few questions... > > First, regarding the prefix, it seemed that this acts as a toggle, > switching from the one codec to the other. But how do I set which me > system uses by default? Or does iconnect always use the high bandwidth one > by default (such that the always switches to the low bandwidth one)? > > Next, I am still struggling to understand the SIP options and what goes > where. Could you please tell me where the format command goes? Is this an > option on the channel? I thing the allow goes in sip.conf. > > Finally, does this have any impact on the problem where the person called > can not be heard? > > Thanks!!! > > Jim > > --On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz > <[EMAIL PROTECTED]> wrote: > > > Jim, > > > > I changed my extensions entry for iconnect to: > > > > exten => _1XX,1,Dial,SIP/[EMAIL PROTECTED] > > > > and my calls work fine with ulaw. I am calling from a linejack card > > with format=ulaw and SIP with allow=ulaw. > > > > Gregg > > > > On Mon, 2003-03-10 at 23:01, Jim Archer wrote: > >> --On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez > >> <[EMAIL PROTECTED]> wrote: > >> > >> > Iconnect uses codecs g723 and g711 that can be configured for each > >> > account (you can change them by the prefix) > >> > >> I tried adding the in front of a number and it reliably generates > >> error "488 invalid media." > >> > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Ok! When I use the prefix and I allow gsm it does work! And the quality is fine. There are two problems we're having now. 1 - From watching the udp fly by, it seems that iconnect does not know when we hang up. For example, if I call a voice mail and it starts giving me its speal and I hang up, iconnect stays connected until the VM hangs up at its end. Next, if we try to call out via iconnect from a sip client extension (like a windows soft phone) all we hear is horrible noise. Has anyone else had these issues? Jim --On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz <[EMAIL PROTECTED]> wrote: I haven't play around enough to know whether or not the prefix is a toggle. I will do some experimenting and let you know. Right now I am prefixing all my calls with . My experience is that when the carrier's format is G723.1, you can't hear the incoming voice. When it is in G711 you can. I have made several calls using G711 and they are acceptable quality. Note that if you disallow=gsm in the sip.conf file you will get the 488 media errors you reported earlier. Below are my config files for sip and the linejack cards: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context=iconnect; Default for incoming calls allow=gsm allow=ulaw allow=alaw ;register=1813342:[EMAIL PROTECTED] ;register=1202454:[EMAIL PROTECTED] [iconnecthere] type=friend username= secret=XXX host=sipauth.deltathree.com ; ; Linux Telephony Interface ; ; Configuration file ; [interfaces] mode=dialtone format=ulaw echocancel=medium silencesupression=no context=local context=default txgain=100% rxgain=100% device => /dev/phone0 On Tue, 2003-03-11 at 14:28, Jim Archer wrote: Hi Greg and thanks very much... A few questions... First, regarding the prefix, it seemed that this acts as a toggle, switching from the one codec to the other. But how do I set which me system uses by default? Or does iconnect always use the high bandwidth one by default (such that the always switches to the low bandwidth one)? Next, I am still struggling to understand the SIP options and what goes where. Could you please tell me where the format command goes? Is this an option on the channel? I thing the allow goes in sip.conf. Finally, does this have any impact on the problem where the person called can not be heard? Thanks!!! Jim --On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz <[EMAIL PROTECTED]> wrote: > Jim, > > I changed my extensions entry for iconnect to: > > exten => _1XX,1,Dial,SIP/[EMAIL PROTECTED] > > and my calls work fine with ulaw. I am calling from a linejack card > with format=ulaw and SIP with allow=ulaw. > > Gregg > > On Mon, 2003-03-10 at 23:01, Jim Archer wrote: >> --On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez >> <[EMAIL PROTECTED]> wrote: >> >> > Iconnect uses codecs g723 and g711 that can be configured for each >> > account (you can change them by the prefix) >> >> I tried adding the in front of a number and it reliably generates >> error "488 invalid media." >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Jim, I am seeing the same hangup problem. The only client I am using with iconnect is their windows dialer. It seems to work well. Gregg On Tue, 2003-03-11 at 18:17, Jim Archer wrote: > Ok! When I use the prefix and I allow gsm it does work! And the > quality is fine. > > There are two problems we're having now. > > 1 - From watching the udp fly by, it seems that iconnect does not know when > we hang up. For example, if I call a voice mail and it starts giving me > its speal and I hang up, iconnect stays connected until the VM hangs up at > its end. > > Next, if we try to call out via iconnect from a sip client extension (like > a windows soft phone) all we hear is horrible noise. > > Has anyone else had these issues? > > Jim > > > --On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz > <[EMAIL PROTECTED]> wrote: > > > I haven't play around enough to know whether or not the prefix is a > > toggle. I will do some experimenting and let you know. Right now I am > > prefixing all my calls with . > > > > My experience is that when the carrier's format is G723.1, you can't > > hear the incoming voice. When it is in G711 you can. I have made several > > calls using G711 and they are acceptable quality. Note that if you > > disallow=gsm in the sip.conf file you will get the 488 media errors you > > reported earlier. > > > > Below are my config files for sip and the linejack cards: > > > > ; > > ; SIP Configuration for Asterisk > > ; > > [general] > > port = 5060 ; Port to bind to > > bindaddr = 0.0.0.0 ; Address to bind to > > context=iconnect; Default for incoming calls > > allow=gsm > > allow=ulaw > > allow=alaw > > > > ;register=1813342:[EMAIL PROTECTED] > > ;register=1202454:[EMAIL PROTECTED] > > > > [iconnecthere] > > type=friend > > username= > > secret=XXX > > host=sipauth.deltathree.com > > > > ; > > ; Linux Telephony Interface > > ; > > ; Configuration file > > ; > > [interfaces] > > > > mode=dialtone > > format=ulaw > > echocancel=medium > > silencesupression=no > > > > context=local > > context=default > > > > txgain=100% > > rxgain=100% > > device => /dev/phone0 > > > > > > > > On Tue, 2003-03-11 at 14:28, Jim Archer wrote: > >> Hi Greg and thanks very much... > >> > >> A few questions... > >> > >> First, regarding the prefix, it seemed that this acts as a toggle, > >> switching from the one codec to the other. But how do I set which me > >> system uses by default? Or does iconnect always use the high bandwidth > >> one by default (such that the always switches to the low bandwidth > >> one)? > >> > >> Next, I am still struggling to understand the SIP options and what goes > >> where. Could you please tell me where the format command goes? Is this > >> an option on the channel? I thing the allow goes in sip.conf. > >> > >> Finally, does this have any impact on the problem where the person > >> called can not be heard? > >> > >> Thanks!!! > >> > >> Jim > >> > >> --On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz > >> <[EMAIL PROTECTED]> wrote: > >> > >> > Jim, > >> > > >> > I changed my extensions entry for iconnect to: > >> > > >> > exten => _1XX,1,Dial,SIP/[EMAIL PROTECTED] > >> > > >> > and my calls work fine with ulaw. I am calling from a linejack card > >> > with format=ulaw and SIP with allow=ulaw. > >> > > >> > Gregg > >> > > >> > On Mon, 2003-03-10 at 23:01, Jim Archer wrote: > >> >> --On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez > >> >> <[EMAIL PROTECTED]> wrote: > >> >> > >> >> > Iconnect uses codecs g723 and g711 that can be configured for each > >> >> > account (you can change them by the prefix) > >> >> > >> >> I tried adding the in front of a number and it reliably generates > >> >> error "488 invalid media." > >> >> > >> >> > >> >> ___ > >> >> Asterisk-Users mailing list > >> >> [EMAIL PROTECTED] > >> >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > ___ > >> > Asterisk-Users mailing list > >> > [EMAIL PROTECTED] > >> > http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
I found similar problems. With my phonejack I can make a call with ulaw with decent quality (I have a 64k line). However, with Messenger I hear a brief horrible noise and that´s it. - Original Message - From: "Jim Archer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 11, 2003 8:17 PM Subject: Re: [Asterisk-Users] iconnect quality? > Ok! When I use the prefix and I allow gsm it does work! And the > quality is fine. > > There are two problems we're having now. > > 1 - From watching the udp fly by, it seems that iconnect does not know when > we hang up. For example, if I call a voice mail and it starts giving me > its speal and I hang up, iconnect stays connected until the VM hangs up at > its end. > > Next, if we try to call out via iconnect from a sip client extension (like > a windows soft phone) all we hear is horrible noise. > > Has anyone else had these issues? > > Jim > > > --On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz > <[EMAIL PROTECTED]> wrote: > > > I haven't play around enough to know whether or not the prefix is a > > toggle. I will do some experimenting and let you know. Right now I am > > prefixing all my calls with . > > > > My experience is that when the carrier's format is G723.1, you can't > > hear the incoming voice. When it is in G711 you can. I have made several > > calls using G711 and they are acceptable quality. Note that if you > > disallow=gsm in the sip.conf file you will get the 488 media errors you > > reported earlier. > > > > Below are my config files for sip and the linejack cards: > > > > ; > > ; SIP Configuration for Asterisk > > ; > > [general] > > port = 5060 ; Port to bind to > > bindaddr = 0.0.0.0 ; Address to bind to > > context=iconnect ; Default for incoming calls > > allow=gsm > > allow=ulaw > > allow=alaw > > > > ;register=1813342:[EMAIL PROTECTED] > > ;register=1202454:[EMAIL PROTECTED] > > > > [iconnecthere] > > type=friend > > username= > > secret=XXX > > host=sipauth.deltathree.com > > > > ; > > ; Linux Telephony Interface > > ; > > ; Configuration file > > ; > > [interfaces] > > > > mode=dialtone > > format=ulaw > > echocancel=medium > > silencesupression=no > > > > context=local > > context=default > > > > txgain=100% > > rxgain=100% > > device => /dev/phone0 > > > > > > > > On Tue, 2003-03-11 at 14:28, Jim Archer wrote: > >> Hi Greg and thanks very much... > >> > >> A few questions... > >> > >> First, regarding the prefix, it seemed that this acts as a toggle, > >> switching from the one codec to the other. But how do I set which me > >> system uses by default? Or does iconnect always use the high bandwidth > >> one by default (such that the always switches to the low bandwidth > >> one)? > >> > >> Next, I am still struggling to understand the SIP options and what goes > >> where. Could you please tell me where the format command goes? Is this > >> an option on the channel? I thing the allow goes in sip.conf. > >> > >> Finally, does this have any impact on the problem where the person > >> called can not be heard? > >> > >> Thanks!!! > >> > >> Jim > >> > >> --On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz > >> <[EMAIL PROTECTED]> wrote: > >> > >> > Jim, > >> > > >> > I changed my extensions entry for iconnect to: > >> > > >> > exten => _1XX,1,Dial,SIP/[EMAIL PROTECTED] > >> > > >> > and my calls work fine with ulaw. I am calling from a linejack card > >> > with format=ulaw and SIP with allow=ulaw. > >> > > >> > Gregg > >> > > >> > On Mon, 2003-03-10 at 23:01, Jim Archer wrote: > >> >> --On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez > >> >> <[EMAIL PROTECTED]> wrote: > >> >> > >> >> > Iconnect uses codecs g723 and g711 that can be configured for each > >> >> > account (you can change them by the prefix) > >> >> > >> >> I tried adding the in front of a number and it reliably generates > >> >> error "488 invalid media." > >> >> > >> >> > >> >> ___
Re: [Asterisk-Users] iconnect quality?
> 1 - From watching the udp fly by, it seems that iconnect does not know > when we hang up. For example, if I call a voice mail and it starts > giving me its speal and I hang up, iconnect stays connected until the VM > hangs up at its end. Because Asterisk doesn't implement RTCP. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
> 1 - From watching the udp fly by, it seems that iconnect does not know when we hang up. For example, if I call a voice mail and it starts giving me its speal and I hang up, iconnect stays connected until the VM hangs up at its end. Because Asterisk doesn't implement RTCP. That should have nothing to do with it, right? If a SIP "BYE" message gets sent to the remote end by Asterisk, the RTP connection should get shut down.Or am I missing something obvious here? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
--On Tuesday, March 11, 2003 5:31 PM -0800 John Todd <[EMAIL PROTECTED]> wrote: Because Asterisk doesn't implement RTCP. That should have nothing to do with it, right? If a SIP "BYE" message gets sent to the remote end by Asterisk, the RTP connection should get shut down.Or am I missing something obvious here? this was my thought as well. There are a number of conditions under which Asterisk does not tell iconnect "BYE." I often see "temporarialy unavailable" and other errors from iconnect, and the Dail app exits, but iconnect keeps on sending the error messages. I think if we hang up on our end we should be able to end the call. I checked the iconnect account page, and I have been billed for lengthy calls that I hung up on right at the beginning. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Actually I think it was an issue with incrementing the sequence number on the bye andshould be fixed now. RTCP is irrelevant in SIP signalling. Mark On Tue, 11 Mar 2003 [EMAIL PROTECTED] wrote: > > 1 - From watching the udp fly by, it seems that iconnect does not know > > when we hang up. For example, if I call a voice mail and it starts > > giving me its speal and I hang up, iconnect stays connected until the VM > > hangs up at its end. > Because Asterisk doesn't implement RTCP. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
IM not sure this is the only cause of the "480 temporarily unavailable" message from Iconnect. I noticed that the dialer from Iconnect put a 20 wait time between calls. Gregg On Tue, 2003-03-11 at 20:53, Jim Archer wrote: > --On Tuesday, March 11, 2003 5:31 PM -0800 John Todd <[EMAIL PROTECTED]> > wrote: > > >> Because Asterisk doesn't implement RTCP. > > > > That should have nothing to do with it, right? If a SIP "BYE" message > > gets sent to the remote end by Asterisk, the RTP connection should get > > shut down.Or am I missing something obvious here? > > this was my thought as well. There are a number of conditions under which > Asterisk does not tell iconnect "BYE." I often see "temporarialy > unavailable" and other errors from iconnect, and the Dail app exits, but > iconnect keeps on sending the error messages. > > I think if we hang up on our end we should be able to end the call. I > checked the iconnect account page, and I have been billed for lengthy calls > that I hung up on right at the beginning. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Dan, why are you using phonejack with ulaw codec? g723 (format=slinear only) is working just perfect with phonejack and iconnect :) Lubo Dan Fernandez wrote: I found similar problems. With my phonejack I can make a call with ulaw with decent quality (I have a 64k line). However, with Messenger I hear a brief horrible noise and thatґs it. - Original Message - From: "Jim Archer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 11, 2003 8:17 PM Subject: Re: [Asterisk-Users] iconnect quality? Ok! When I use the prefix and I allow gsm it does work! And the quality is fine. There are two problems we're having now. 1 - From watching the udp fly by, it seems that iconnect does not know when we hang up. For example, if I call a voice mail and it starts giving me its speal and I hang up, iconnect stays connected until the VM hangs up at its end. Next, if we try to call out via iconnect from a sip client extension (like a windows soft phone) all we hear is horrible noise. Has anyone else had these issues? Jim --On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz <[EMAIL PROTECTED]> wrote: I haven't play around enough to know whether or not the prefix is a toggle. I will do some experimenting and let you know. Right now I am prefixing all my calls with . My experience is that when the carrier's format is G723.1, you can't hear the incoming voice. When it is in G711 you can. I have made several calls using G711 and they are acceptable quality. Note that if you disallow=gsm in the sip.conf file you will get the 488 media errors you reported earlier. Below are my config files for sip and the linejack cards: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context=iconnect ; Default for incoming calls allow=gsm allow=ulaw allow=alaw ;register=1813342:[EMAIL PROTECTED] ;register=1202454:[EMAIL PROTECTED] [iconnecthere] type=friend username= secret=XXX host=sipauth.deltathree.com ; ; Linux Telephony Interface ; ; Configuration file ; [interfaces] mode=dialtone format=ulaw echocancel=medium silencesupression=no context=local context=default txgain=100% rxgain=100% device => /dev/phone0 On Tue, 2003-03-11 at 14:28, Jim Archer wrote: Hi Greg and thanks very much... A few questions... First, regarding the prefix, it seemed that this acts as a toggle, switching from the one codec to the other. But how do I set which me system uses by default? Or does iconnect always use the high bandwidth one by default (such that the always switches to the low bandwidth one)? Next, I am still struggling to understand the SIP options and what goes where. Could you please tell me where the format command goes? Is this an option on the channel? I thing the allow goes in sip.conf. Finally, does this have any impact on the problem where the person called can not be heard? Thanks!!! Jim --On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz <[EMAIL PROTECTED]> wrote: Jim, I changed my extensions entry for iconnect to: exten => _1XX,1,Dial,SIP/[EMAIL PROTECTED] and my calls work fine with ulaw. I am calling from a linejack card with format=ulaw and SIP with allow=ulaw. Gregg On Mon, 2003-03-10 at 23:01, Jim Archer wrote: --On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez <[EMAIL PROTECTED]> wrote: Iconnect uses codecs g723 and g711 that can be configured for each account (you can change them by the prefix) I tried adding the in front of a number and it reliably generates error "488 invalid media." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
I pulled the CVS tree at 11:30PM (approximately) Eastern on March 11. This problem does seem to be fixed. But there is a new issue. It now seems that adding the prefix that solved the problems before now causes the "488 Not Acceptable Media" error response: 856.137213 192.203.175.9 -> 213.137.73.178 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 856.283972 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 856.284294 213.137.73.176 -> 192.203.175.9 SIP Status: 407 Proxy Authentication Required 856.284651 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 856.285463 192.203.175.9 -> 213.137.73.178 IP Fragmented IP protocol (proto=UDP 0x11, off=552) 856.285493 192.203.175.9 -> 213.137.73.178 SIP Request: INVITE sip:[EMAIL PROTECTED] 856.423732 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 856.594346 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 856.598273 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 857.233615 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 858.123355 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 861.142317 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 867.170217 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media Removing the prefix seems to make an improvement. The called phone will ring. If I answer it, though, Asterisk does not seem to know and keeps ringing my phone. If I call my cell phone and hang up Asterisk seems to notice, but if I call a POTS line, pick it up then hang up Asterisj seems not to notice: 56.265536 192.203.175.9 -> 213.137.73.178 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 56.390918 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 56.391242 213.137.73.176 -> 192.203.175.9 SIP Status: 407 Proxy Authentication Required 56.391571 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 56.392423 192.203.175.9 -> 213.137.73.178 IP Fragmented IP protocol (proto=UDP 0x11, off=552) 56.392453 192.203.175.9 -> 213.137.73.178 SIP Request: INVITE sip:[EMAIL PROTECTED] 56.521701 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 59.769661 213.137.73.176 -> 192.203.175.9 SIP Status: 180 Ringing 64.637854 213.137.73.176 -> 192.203.175.9 SIP Status: 404 Not Found 78.543317 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 78.544346 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 80.062729 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 83.111677 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 88.659719 213.137.73.176 -> 192.203.175.9 SIP Status: 404 Not Found Another example... 85.770587 192.203.175.9 -> 213.137.73.178 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 85.897768 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 85.898134 213.137.73.176 -> 192.203.175.9 SIP Status: 407 Proxy Authentication Required 85.898581 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 85.899266 192.203.175.9 -> 213.137.73.178 IP Fragmented IP protocol (proto=UDP 0x11, off=552) 85.899294 192.203.175.9 -> 213.137.73.178 SIP Request: INVITE sip:[EMAIL PROTECTED] 86.097458 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 89.116528 213.137.73.176 -> 192.203.175.9 SIP Status: 180 Ringing 100.312843 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 100.313648 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 101.812307 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 104.851267 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 110.899191 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media --On Tuesday, March 11, 2003 8:04 PM -0600 Mark Spencer <[EMAIL PROTECTED]> wrote: Actually I think it was an issue with incrementing the sequence number on the bye andshould be fixed now. RTCP is irrelevant in SIP signalling. Mark On Tue, 11 Mar 2003 [EMAIL PROTECTED] wrote: > 1 - From watching the udp fly by, it seems that iconnect does not know > when we hang up. For example, if I call a voice mail and it starts > giving me its speal and I hang up, iconnect stays connected until the > VM hangs up at its end. Because Asterisk doesn't implement RTCP. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Hi Lubo, I appreciate your email to help with this issue, but I don't understand your message. I assume your comment about format=slinear is to use format=slinear in phone.conf instead of format=ulaw. If so, how does this get you g723 to iconnect? Using format=g723.1 doesn't seem to work. Gregg On Wed, 2003-03-12 at 00:50, Lubomir Christov wrote: > Dan, why are you using phonejack with ulaw codec? g723 (format=slinear > only) is working just perfect with phonejack and iconnect :) > > Lubo > > Dan Fernandez wrote: > > I found similar problems. > > > > With my phonejack I can make a call with ulaw with decent quality (I have a > > 64k line). > > > > However, with Messenger I hear a brief horrible noise and thatґs it. > > > > - Original Message - > > From: "Jim Archer" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Tuesday, March 11, 2003 8:17 PM > > Subject: Re: [Asterisk-Users] iconnect quality? > > > > > > > >>Ok! When I use the prefix and I allow gsm it does work! And the > >>quality is fine. > >> > >>There are two problems we're having now. > >> > >>1 - From watching the udp fly by, it seems that iconnect does not know > > > > when > > > >>we hang up. For example, if I call a voice mail and it starts giving me > >>its speal and I hang up, iconnect stays connected until the VM hangs up at > >>its end. > >> > >>Next, if we try to call out via iconnect from a sip client extension (like > >>a windows soft phone) all we hear is horrible noise. > >> > >>Has anyone else had these issues? > >> > >>Jim > >> > >> > >>--On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz > >><[EMAIL PROTECTED]> wrote: > >> > >> > >>>I haven't play around enough to know whether or not the prefix is a > >>>toggle. I will do some experimenting and let you know. Right now I am > >>>prefixing all my calls with . > >>> > >>>My experience is that when the carrier's format is G723.1, you can't > >>>hear the incoming voice. When it is in G711 you can. I have made several > >>>calls using G711 and they are acceptable quality. Note that if you > >>>disallow=gsm in the sip.conf file you will get the 488 media errors you > >>>reported earlier. > >>> > >>>Below are my config files for sip and the linejack cards: > >>> > >>>; > >>>; SIP Configuration for Asterisk > >>>; > >>>[general] > >>>port = 5060 ; Port to bind to > >>>bindaddr = 0.0.0.0 ; Address to bind to > >>>context=iconnect ; Default for incoming calls > >>>allow=gsm > >>>allow=ulaw > >>>allow=alaw > >>> > >>>;register=1813342:[EMAIL PROTECTED] > >>>;register=1202454:[EMAIL PROTECTED] > >>> > >>>[iconnecthere] > >>>type=friend > >>>username= > >>>secret=XXX > >>>host=sipauth.deltathree.com > >>> > >>>; > >>>; Linux Telephony Interface > >>>; > >>>; Configuration file > >>>; > >>>[interfaces] > >>> > >>>mode=dialtone > >>>format=ulaw > >>>echocancel=medium > >>>silencesupression=no > >>> > >>>context=local > >>>context=default > >>> > >>>txgain=100% > >>>rxgain=100% > >>>device => /dev/phone0 > >>> > >>> > >>> > >>>On Tue, 2003-03-11 at 14:28, Jim Archer wrote: > >>> > >>>>Hi Greg and thanks very much... > >>>> > >>>>A few questions... > >>>> > >>>>First, regarding the prefix, it seemed that this acts as a toggle, > >>>>switching from the one codec to the other. But how do I set which me > >>>>system uses by default? Or does iconnect always use the high bandwidth > >>>>one by default (such that the always switches to the low > > > > bandwidth > > > >>>>one)? > >>>> > >>>>Next, I am still struggling to understand the SIP options and what goes > >>>>where. Could you please tell me where the format command goes? Is > > > > this > > > >>>>an opti
Re: [Asterisk-Users] iconnect quality?
Hi Gregg,
I'm using iconnect with LineJACK/PhoneCARD and G723.1 codec from about 1
mount without any problems. The quality is perfect and everything is OK
(only some little problems sometime).
But today morning, with the NEW CVS version update of asterisk I found
that SIP(G723/ulaw) and iconnect aren't working anymore ???
When I try to connect trough iconnect I receive this error message:
-- Got SIP response 488 "Not Acceptable Media" back from 213.137.73.178
you can try asterisk from yesterday:
cvs -z9 co -D "Mar 11 2003" asterisk
and test it: everything will be OK :)
Here is my working configuration:
sip.conf
[general]
port = 5060
;bindaddr = 0.0.0.0
context = incomming
disallow=all
allow=g723.1
;allow=ulaw
tos=lowdelay
tos=184
[iconnect]
type=friend
username=12345678
password=1234
host=213.137.73.178
callerid=1234567890
phone.conf
format=slinear
echocancel=low
silencesupression=no
extension.conf
exten => _00.,1,Dial(Sip/${EXTEN:[EMAIL PROTECTED],,C)
Lubo
P.S. for successfully using G723 codec and phonejack you will need
g723.1 and g723.1b placed in your codecs directory. You can got it like
this:
cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co g723.1
cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co g723.1b
and uncomment this line in Makefile in codecs directory
MODG723=codec_g723_1.so codec_g723_1b.so
I hope that the todays problem with asterisk and SIP/G723 will be fixed
very soon.
L
Gregg Lebovitz wrote:
Hi Lubo,
I appreciate your email to help with this issue, but I don't understand
your message. I assume your comment about format=slinear is to use
format=slinear in phone.conf instead of format=ulaw. If so, how does
this get you g723 to iconnect? Using format=g723.1 doesn't seem to work.
Gregg
On Wed, 2003-03-12 at 00:50, Lubomir Christov wrote:
Dan, why are you using phonejack with ulaw codec? g723 (format=slinear
only) is working just perfect with phonejack and iconnect :)
Lubo
Dan Fernandez wrote:
I found similar problems.
With my phonejack I can make a call with ulaw with decent quality (I have a
64k line).
However, with Messenger I hear a brief horrible noise and that?s it.
- Original Message -
From: "Jim Archer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, March 11, 2003 8:17 PM
Subject: Re: [Asterisk-Users] iconnect quality?
Ok! When I use the prefix and I allow gsm it does work! And the
quality is fine.
There are two problems we're having now.
1 - From watching the udp fly by, it seems that iconnect does not know
when
we hang up. For example, if I call a voice mail and it starts giving me
its speal and I hang up, iconnect stays connected until the VM hangs up at
its end.
Next, if we try to call out via iconnect from a sip client extension (like
a windows soft phone) all we hear is horrible noise.
Has anyone else had these issues?
Jim
--On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz
<[EMAIL PROTECTED]> wrote:
I haven't play around enough to know whether or not the prefix is a
toggle. I will do some experimenting and let you know. Right now I am
prefixing all my calls with .
My experience is that when the carrier's format is G723.1, you can't
hear the incoming voice. When it is in G711 you can. I have made several
calls using G711 and they are acceptable quality. Note that if you
disallow=gsm in the sip.conf file you will get the 488 media errors you
reported earlier.
Below are my config files for sip and the linejack cards:
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context=iconnect ; Default for incoming calls
allow=gsm
allow=ulaw
allow=alaw
;register=1813342:[EMAIL PROTECTED]
;register=1202454:[EMAIL PROTECTED]
[iconnecthere]
type=friend
username=
secret=XXX
host=sipauth.deltathree.com
;
; Linux Telephony Interface
;
; Configuration file
;
[interfaces]
mode=dialtone
format=ulaw
echocancel=medium
silencesupression=no
context=local
context=default
txgain=100%
rxgain=100%
device => /dev/phone0
On Tue, 2003-03-11 at 14:28, Jim Archer wrote:
Hi Greg and thanks very much...
A few questions...
First, regarding the prefix, it seemed that this acts as a toggle,
switching from the one codec to the other. But how do I set which me
system uses by default? Or does iconnect always use the high bandwidth
one by default (such that the always switches to the low
bandwidth
one)?
Next, I am still struggling to understand the SIP options and what goes
where. Could you please tell me where the format command goes? Is
this
an option on the channel? I thing the allow goes in sip.conf.
Finally, does this have any impact on the problem where the person
called can not be heard?
Thanks!!!
Jim
--On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz
<[EMAIL P
Re: [Asterisk-Users] iconnect quality?
Please check latest CVS. This issue has been fixed and was related to the
dynamic payload merger.
Mark
On Wed, 12 Mar 2003, Lubomir Christov wrote:
> Hi Gregg,
>
> I'm using iconnect with LineJACK/PhoneCARD and G723.1 codec from about 1
> mount without any problems. The quality is perfect and everything is OK
> (only some little problems sometime).
> But today morning, with the NEW CVS version update of asterisk I found
> that SIP(G723/ulaw) and iconnect aren't working anymore ???
> When I try to connect trough iconnect I receive this error message:
>
> -- Got SIP response 488 "Not Acceptable Media" back from 213.137.73.178
>
> you can try asterisk from yesterday:
>cvs -z9 co -D "Mar 11 2003" asterisk
>
> and test it: everything will be OK :)
> Here is my working configuration:
>
> sip.conf
>
> [general]
> port = 5060
> ;bindaddr = 0.0.0.0
> context = incomming
> disallow=all
> allow=g723.1
> ;allow=ulaw
> tos=lowdelay
> tos=184
>
> [iconnect]
> type=friend
> username=12345678
> password=1234
> host=213.137.73.178
> callerid=1234567890
>
>
>
> phone.conf
>
> format=slinear
> echocancel=low
> silencesupression=no
>
>
> extension.conf
>
> exten => _00.,1,Dial(Sip/${EXTEN:[EMAIL PROTECTED],,C)
>
> Lubo
>
> P.S. for successfully using G723 codec and phonejack you will need
> g723.1 and g723.1b placed in your codecs directory. You can got it like
> this:
>
> cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co g723.1
> cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co g723.1b
>
> and uncomment this line in Makefile in codecs directory
> MODG723=codec_g723_1.so codec_g723_1b.so
>
> I hope that the todays problem with asterisk and SIP/G723 will be fixed
> very soon.
>
> L
>
> Gregg Lebovitz wrote:
> > Hi Lubo,
> >
> > I appreciate your email to help with this issue, but I don't understand
> > your message. I assume your comment about format=slinear is to use
> > format=slinear in phone.conf instead of format=ulaw. If so, how does
> > this get you g723 to iconnect? Using format=g723.1 doesn't seem to work.
> >
> > Gregg
> >
> > On Wed, 2003-03-12 at 00:50, Lubomir Christov wrote:
> >
> >>Dan, why are you using phonejack with ulaw codec? g723 (format=slinear
> >>only) is working just perfect with phonejack and iconnect :)
> >>
> >>Lubo
> >>
> >>Dan Fernandez wrote:
> >>
> >>>I found similar problems.
> >>>
> >>>With my phonejack I can make a call with ulaw with decent quality (I have a
> >>>64k line).
> >>>
> >>>However, with Messenger I hear a brief horrible noise and thats it.
> >>>
> >>>- Original Message -
> >>>From: "Jim Archer" <[EMAIL PROTECTED]>
> >>>To: <[EMAIL PROTECTED]>
> >>>Sent: Tuesday, March 11, 2003 8:17 PM
> >>>Subject: Re: [Asterisk-Users] iconnect quality?
> >>>
> >>>
> >>>
> >>>
> >>>>Ok! When I use the prefix and I allow gsm it does work! And the
> >>>>quality is fine.
> >>>>
> >>>>There are two problems we're having now.
> >>>>
> >>>>1 - From watching the udp fly by, it seems that iconnect does not know
> >>>
> >>>when
> >>>
> >>>
> >>>>we hang up. For example, if I call a voice mail and it starts giving me
> >>>>its speal and I hang up, iconnect stays connected until the VM hangs up at
> >>>>its end.
> >>>>
> >>>>Next, if we try to call out via iconnect from a sip client extension (like
> >>>>a windows soft phone) all we hear is horrible noise.
> >>>>
> >>>>Has anyone else had these issues?
> >>>>
> >>>>Jim
> >>>>
> >>>>
> >>>>--On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz
> >>>><[EMAIL PROTECTED]> wrote:
> >>>>
> >>>>
> >>>>
> >>>>>I haven't play around enough to know whether or not the prefix is a
> >>>>>toggle. I will do some experimenting and let you know. Right now I am
> >>>>>prefixing all my calls with .
> >>>>>
> >>>>>My experience is that when the carrier's format is G723.1, you can't
> >>>>>hear the incoming voice. When it is in G71
