hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved useragent "SJLabs-SJphone/1.40.258" for peer 5000 [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no [5000] type=friend port=5060 canreinvite=no host=dynamic nat=yes insecure=yes auth=plaintext ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs
At firewall/NAT you have to do port forwarding. If your phone is at port 5060, NAT device will receive a connection and has to know that it is destined for your SIP phone. So, forward port 5060 to the phone. Rudolf ----- Original Message ----- From: "Kamran Ahmad" <p_kami@yahoo.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, August 14, 2005 6:52 AM Subject: [Asterisk-Users] Why NAT problem> hello > > i am using asterisk-1.0.9. i have a NAT problem. > without NAT registration is ok. and if user is bhind > NAT it is registring on asterisk. but SJPhone is > showing "not registered". i think asterisk is properly > sending request to UA. any comments............this > sip.conf setting was working previously > > -- Registered SIP '5000' at 0.0.0.0 port 5060 > expires 120 > -- Saved useragent "SJLabs-SJphone/1.40.258" for > peer 5000 > > [general] > context=default > port=5060 > bindaddr=0.0.0.0 > srvlookup=yes > nat=yes > canreinvite=no > > [5000] > type=friend > port=5060 > canreinvite=no > host=dynamic > nat=yes > insecure=yes > auth=plaintext > > > > > > ____________________________________________________ > Start your day with Yahoo! - make it your home page > http://www.yahoo.com/r/hs > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Rudolf Ladyzhenskii
2005-Aug-13 18:57 UTC
[Asterisk-Users] Echo problem -- network related?
Hi, all I am running asterisk and my friends are running FireFly IAX phone. All is fine except one of them. When anyone tries to talk to him, tehre is a real bad echo. It is nothing to do with sound setup. For example, I call him and I can hear myself speaking with some delay. In "lame terms" it feels like my voice is bouncing off his computer and I can hear myself very clearly. At same time, on his end my voice is heard but it is breaking up. I have another friend using same ISP and there are no problems. I also checked his PC for mailware and other software that can impact on networking performance. All seems OK. I am going to take his PC and connect it on same LAN astersik is sitting on to make sure it is not PC doing it. In the meanwile I would appreciate any ideas you might have. Thanks, Rudolf
Rudolf Ladyzhenskii wrote:> Hi, all > > I am running asterisk and my friends are running FireFly IAX phone. All > is fine except one of them. When anyone tries to talk to him, tehre is > a real bad echo. It is nothing to do with sound setup.Is he using a headset or speakers and microphone? Does he have Stereo Mix selected as a recording source? -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
Rudolf Ladyzhenskii
2005-Aug-13 19:39 UTC
[Asterisk-Users] Echo problem -- network related?
Hi, The problem is not sound setup related. It present even if microphone is disconnected. Rudolf ----- Original Message ----- From: "Matt Riddell" <matt.riddell@sineapps.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, August 14, 2005 12:12 PM Subject: Re: [Asterisk-Users] Echo problem -- network related?> Rudolf Ladyzhenskii wrote: >> Hi, all >> >> I am running asterisk and my friends are running FireFly IAX phone. All >> is fine except one of them. When anyone tries to talk to him, tehre is >> a real bad echo. It is nothing to do with sound setup. > > Is he using a headset or speakers and microphone? > > Does he have Stereo Mix selected as a recording source? > > -- > Cheers, > > Matt Riddell > _______________________________________________ > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Rudolf Ladyzhenskii
2005-Aug-14 00:04 UTC
[Asterisk-Users] Echo problem -- network related?
Hi, I am using SIP phone (Polycom 300). Echo is present even if other party has sound hardware disconnected. It is definetely network and/or PC setup issue, but is not related to audio setup. I will check stereo mix, however. Rudolf ----- Original Message ----- From: "Peter Svensson" <psvasterisk@psv.nu> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, August 14, 2005 5:05 PM Subject: Re: [Asterisk-Users] Echo problem -- network related?> On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: > >> The problem is not sound setup related. It present even if microphone is >> disconnected. > > To repeat the question from Matt Riddell: > >> > Does he have Stereo Mix selected as a recording source? > > We have found the most common cause of a strong echo to be that the sound > card is set to record the outgoing earphone signal. > > If you post inline it is much easier to see what your answers were to > different questions or if you have missed one. > > Peter > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote:> The problem is not sound setup related. It present even if microphone is > disconnected.To repeat the question from Matt Riddell:> > Does he have Stereo Mix selected as a recording source?We have found the most common cause of a strong echo to be that the sound card is set to record the outgoing earphone signal. If you post inline it is much easier to see what your answers were to different questions or if you have missed one. Peter
Geoff wrote:>Hi, Does anyone know the optimum sound quality to use for playing recorded >voice prompts with Asterisk? What is the best format, quality, etc? > > > >http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files Doug
Thanks Doug. Perfect. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Doug Lytle Sent: Sunday, 14 August 2005 10:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sound Quality? Geoff wrote:>Hi, Does anyone know the optimum sound quality to use for playing >recorded voice prompts with Asterisk? What is the best format, quality,etc?> > > >http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files Doug _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
hello but this conf was working for me when i installed asterisk last time. and UA was successfully reg and working. it is a simple case UA behind NAT and Asterisk is on public ip [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no [5000] type=friend port=5060 canreinvite=no host=dynamic nat=yes insecure=yes auth=plaintext>In case of IAX phones, this is not appliacble. IAXuses same port for>both >control and voice. > >I have not tried CISCO phones, but I beleive you doneed port>forwarding if >they are SIP phones. Otherwise, they will not acceptcalls. At least>this is >the case with Polycom phones. > >Rudolf > >----- Original Message ----- >From: "Tom Rymes" <trymes@rymesheating.com> >To: "Asterisk Users Mailing List - Non-CommercialDiscussion"><asterisk-users@lists.digium.com> >Sent: Sunday, August 14, 2005 12:30 PM >Subject: RE: [Asterisk-Users] Why NAT problem > > >> As a followup to my own post, AFAIK, my commentsapply to SIP>clients, >> but you always have to forward the ports to theasterisk server...>> >>> -----Original Message----- >>> From: asterisk-users-bounces@lists.digium.com >>> [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of>>> Tom Rymes >>> Sent: Saturday, August 13, 2005 10:22 PM >>> To: Asterisk Users Mailing List - Non-CommercialDiscussion>>> Subject: RE: [Asterisk-Users] Why NAT problem >>> >>> >>> This is not technically true. For instance, youcan take a>>> Cisco 79X0 and put it behind NAT and it will workwithout>>> port forwarding. You do, however, have to programthe phone>>> to enable the NAT features. (There are two, Ican't remember>>> their names, though.) I have generally left theWAN IP>>> address blank, with no noticable ill effects, butthat might>>> not be a good idea. >>> >>> Also, I believe that you can do this with multiplephones, so>>> long as you use different port numbers for eachphone (5061,>>> 5062, etc) >>> >>> Tom >>> >>> > -----Original Message----- >>> > From: asterisk-users-bounces@lists.digium.com >>> > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of>>> > Rudolf Ladyzhenskii >>> > Sent: Saturday, August 13, 2005 9:53 PM >>> > To: Asterisk Users Mailing List - Non-CommercialDiscussion>>> > Subject: Re: [Asterisk-Users] Why NAT problem >>> > >>> > >>> > At firewall/NAT you have to do port forwarding. >>> > >>> > If your phone is at port 5060, NAT device willreceive a>>> > connection and has >>> > to know that it is destined for your SIP phone.So, forward>>> > port 5060 to the >>> > phone. >>> > >>> > Rudolf >>> > >>> > >>> > ----- Original Message ----- >>> > From: "Kamran Ahmad" <p_kami@yahoo.com> >>> > To: <asterisk-users@lists.digium.com> >>> > Sent: Sunday, August 14, 2005 6:52 AM >>> > Subject: [Asterisk-Users] Why NAT problem >>> > >>> > >>> > > hello >>> > > >>> > > i am using asterisk-1.0.9. i have a NATproblem.>>> > > without NAT registration is ok. and if user isbhind>>> > > NAT it is registring on asterisk. but SJPhoneis>>> > > showing "not registered". i think asterisk isproperly>>> > sending request >>> > > to UA. any comments............this sip.confsetting was working>>> > > previously >>> > > >>> > > -- Registered SIP '5000' at 0.0.0.0 port5060>>> > > expires 120 >>> > > -- Saved useragent"SJLabs-SJphone/1.40.258" for>>> > > peer 5000 >>> > > >>> > > [general] >>> > > context=default >>> > > port=5060 >>> > > bindaddr=0.0.0.0 >>> > > srvlookup=yes >>> > > nat=yes >>> > > canreinvite=no >>> > > >>> > > [5000] >>> > > type=friend >>> > > port=5060 >>> > > canreinvite=no >>> > > host=dynamic >>> > > nat=yes >>> > > insecure=yes >>> > > auth=plaintext >>> > > >>> > > >>> > > >>> > > >>> > > >>> > >____________________________________________________>>> > > Start your day with Yahoo! - make it your homepage>>> > > http://www.yahoo.com/r/hs >>> > > >>> > >_______________________________________________>>> > > Asterisk-Users mailing list >>> > > Asterisk-Users@lists.digium.com >>> > >http://lists.digium.com/mailman/listinfo/asterisk-users>>> > > To UNSUBSCRIBE or update options visit: >>> > >http://lists.digium.com/mailman/listinfo/asterisk-users>>> > >>> > _______________________________________________ >>> > Asterisk-Users mailing list >>> > Asterisk-Users@lists.digium.com >>> >http://lists.digium.com/mailman/listinfo/asterisk-users>>> > To UNSUBSCRIBE or update options visit: >>> >http://lists.digium.com/mailman/listinfo/asterisk-users>>> > >>> >>> >>> >>> _______________________________________________ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>>http://lists.digium.com/mailman/listinfo/asterisk-users>>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users>>> >> >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users>> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users __________________________________________________ Do You Yahoo!? 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hello but this conf was working for me when i installed asterisk last time. and UA was successfully reg and working. i think port forwarding is not the solution. because it was working without port forwarding in my last installation. it is a simple case UA behind NAT and Asterisk is on public ip [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no [5000] type=friend port=5060 canreinvite=no host=dynamic nat=yes insecure=yes auth=plaintext>In case of IAX phones, this is not appliacble. IAXuses same port for>both >control and voice. > >I have not tried CISCO phones, but I beleive you doneed port>forwarding if >they are SIP phones. Otherwise, they will not acceptcalls. At least>this is >the case with Polycom phones. > >Rudolf > >----- Original Message ----- >From: "Tom Rymes" <trymes@rymesheating.com> >To: "Asterisk Users Mailing List - Non-CommercialDiscussion"><asterisk-users@lists.digium.com> >Sent: Sunday, August 14, 2005 12:30 PM >Subject: RE: [Asterisk-Users] Why NAT problem > > >> As a followup to my own post, AFAIK, my commentsapply to SIP>clients, >> but you always have to forward the ports to theasterisk server...>> >>> -----Original Message----- >>> From: asterisk-users-bounces@lists.digium.com >>> [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of>>> Tom Rymes >>> Sent: Saturday, August 13, 2005 10:22 PM >>> To: Asterisk Users Mailing List - Non-CommercialDiscussion>>> Subject: RE: [Asterisk-Users] Why NAT problem >>> >>> >>> This is not technically true. For instance, youcan take a>>> Cisco 79X0 and put it behind NAT and it will workwithout>>> port forwarding. You do, however, have to programthe phone>>> to enable the NAT features. (There are two, Ican't remember>>> their names, though.) I have generally left theWAN IP>>> address blank, with no noticable ill effects, butthat might>>> not be a good idea. >>> >>> Also, I believe that you can do this with multiplephones, so>>> long as you use different port numbers for eachphone (5061,>>> 5062, etc) >>> >>> Tom >>> >>> > -----Original Message----- >>> > From: asterisk-users-bounces@lists.digium.com >>> > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of>>> > Rudolf Ladyzhenskii >>> > Sent: Saturday, August 13, 2005 9:53 PM >>> > To: Asterisk Users Mailing List - Non-CommercialDiscussion>>> > Subject: Re: [Asterisk-Users] Why NAT problem >>> > >>> > >>> > At firewall/NAT you have to do port forwarding. >>> > >>> > If your phone is at port 5060, NAT device willreceive a>>> > connection and has >>> > to know that it is destined for your SIP phone.So, forward>>> > port 5060 to the >>> > phone. >>> > >>> > Rudolf >>> > >>> > >>> > ----- Original Message ----- >>> > From: "Kamran Ahmad" <p_kami@yahoo.com> >>> > To: <asterisk-users@lists.digium.com> >>> > Sent: Sunday, August 14, 2005 6:52 AM >>> > Subject: [Asterisk-Users] Why NAT problem >>> > >>> > >>> > > hello >>> > > >>> > > i am using asterisk-1.0.9. i have a NATproblem.>>> > > without NAT registration is ok. and if user isbhind>>> > > NAT it is registring on asterisk. but SJPhoneis>>> > > showing "not registered". i think asterisk isproperly>>> > sending request >>> > > to UA. any comments............this sip.confsetting was working>>> > > previously >>> > > >>> > > -- Registered SIP '5000' at 0.0.0.0 port5060>>> > > expires 120 >>> > > -- Saved useragent"SJLabs-SJphone/1.40.258" for>>> > > peer 5000 >>> > > >>> > > [general] >>> > > context=default >>> > > port=5060 >>> > > bindaddr=0.0.0.0 >>> > > srvlookup=yes >>> > > nat=yes >>> > > canreinvite=no >>> > > >>> > > [5000] >>> > > type=friend >>> > > port=5060 >>> > > canreinvite=no >>> > > host=dynamic >>> > > nat=yes >>> > > insecure=yes >>> > > auth=plaintext >>> > > >>> > > >>> > > >>> > > >>> > > >>> > >____________________________________________________>>> > > Start your day with Yahoo! - make it your homepage>>> > > http://www.yahoo.com/r/hs >>> > > >>> > >_______________________________________________>>> > > Asterisk-Users mailing list >>> > > Asterisk-Users@lists.digium.com >>> > >http://lists.digium.com/mailman/listinfo/asterisk-users>>> > > To UNSUBSCRIBE or update options visit: >>> > >http://lists.digium.com/mailman/listinfo/asterisk-users>>> > >>> > _______________________________________________ >>> > Asterisk-Users mailing list >>> > Asterisk-Users@lists.digium.com >>> >http://lists.digium.com/mailman/listinfo/asterisk-users>>> > To UNSUBSCRIBE or update options visit: >>> >http://lists.digium.com/mailman/listinfo/asterisk-users>>> > >>> >>> >>> >>> _______________________________________________ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>>http://lists.digium.com/mailman/listinfo/asterisk-users>>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users>>> >> >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >>____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs
Hi, I had configured Asterisk with the following 1). X100P - Card 2). Two -Greadstream100 SIP Phones. I am able to make calls from SIP Ext to SIP Ext and PSTN calls from outside to SIP Extn. But I am not able to make calls from SIP Extn to PSTN out going calls-it gives BT error message- The number you had dialled not recognised. The SIP extn is not sending the correct number. I will be thank full if some solutions is suggested. appan kh
Hi, I had configured Asterisk with the following 1). X100P - Card 2). Two -Greadstream100 SIP Phones. I am able to make calls from SIP Ext to SIP Ext and PSTN calls from outside to SIP Extn. But I am not able to make calls from SIP Extn to PSTN out going calls-it gives BT error message- The number you had dialled not recognised. The SIP extn is not sending the correct number. I will be thank full if some solutions is suggested. appan kh
Hello, I have read your email. I found that you have configured X100P card and established a call from SIP exten. to SIP exten and PSTN to SIP exten. I have done the first part i.e. SIP exten to SIP exten and would like to do a second part. So please help me regarding this. I have installed Asterisk on Linux machine. So from here please guide me how i should proceed. What are the requirements? and some other details. Your help will be much appriciated. Thanks, Nil. Appan KH <appan@softswitches.net> wrote: Hi, I had configured Asterisk with the following 1). X100P - Card 2). Two -Greadstream100 SIP Phones. I am able to make calls from SIP Ext to SIP Ext and PSTN calls from outside to SIP Extn. But I am not able to make calls from SIP Extn to PSTN out going calls-it gives BT error message- The number you had dialled not recognised. The SIP extn is not sending the correct number. I will be thank full if some solutions is suggested. appan kh _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050923/a8912458/attachment.htm
Hello, I have read your email. I found that you have configured X100P card and established a call from SIP exten. to SIP exten and PSTN to SIP exten. I have done the first part i.e. SIP exten to SIP exten and would like to do a second part. So please help me regarding this. I have installed Asterisk on Linux machine. So from here please guide me how i should proceed. What are the requirements? and some other details. Your help will be much appriciated. Thanks, Nil. Appan KH <appan@softswitches.net> wrote: Hi, I had configured Asterisk with the following 1). X100P - Card 2). Two -Greadstream100 SIP Phones. I am able to make calls from SIP Ext to SIP Ext and PSTN calls from outside to SIP Extn. But I am not able to make calls from SIP Extn to PSTN out going calls-it gives BT error message- The number you had dialled not recognised. The SIP extn is not sending the correct number. I will be thank full if some solutions is suggested. appan kh _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050923/ac2bb4af/attachment.htm