Leandro Dardini
2012-Mar-25 08:56 UTC
[asterisk-users] Routing premature media to the calling channel
Hello, I have a problem with premature media and inband progress audio. I am using the latest 1.8.10.1 and this is the setup: soft phone --- asterisk --- SIP provider The number I call is giving back some hints via inband audio I am not able to ear from the soft phone. They stop on the asterisk and are not routed down the path to the sip phone. The SIP part is simple: soft phone -> asterisk: INVITE asterisk -> soft phone: TRYING asterisk -> provider: INVITE asterisk -> soft phone: 180 RINGING provider -> asterisk: 183 SESSION PROGRESS provider -> asterisk: AUDIO Unfortunately the AUDIO received from the provider by the asterisk box is not sent to the soft phone. I think I have tried every combination of progressinband and prematuremedia, without success. How can I made the audio received from the provider to the asterisk be transmitted to the soft phone? Thank you Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120325/f8adb536/attachment.htm>
Alex Balashov
2012-Mar-25 09:07 UTC
[asterisk-users] Routing premature media to the calling channel
I assume you have ruled out NAT and firewall issues? Between those two, 99% of the reasons why something may not be routed somewhere correctly are accounted for. If you don't know, your best bet is to take a packet capture or SIP debug on the Asterisk server and find out where that early media is going. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini <ldardini at gmail.com> wrote:>Hello, >I have a problem with premature media and inband progress audio. I am using >the latest 1.8.10.1 and this is the setup: > >soft phone --- asterisk --- SIP provider > >The number I call is giving back some hints via inband audio I am not able >to ear from the soft phone. They stop on the asterisk and are not routed >down the path to the sip phone. > >The SIP part is simple: > >soft phone -> asterisk: INVITE > >asterisk -> soft phone: TRYING > >asterisk -> provider: INVITE > >asterisk -> soft phone: 180 RINGING > >provider -> asterisk: 183 SESSION PROGRESS > >provider -> asterisk: AUDIO > >Unfortunately the AUDIO received from the provider by the asterisk box is >not sent to the soft phone. > >I think I have tried every combination of progressinband and >prematuremedia, without success. > >How can I made the audio received from the provider to the asterisk be >transmitted to the soft phone? > >Thank you > >Leandro > >-- >_____________________________________________________________________ >-- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120325/be1a3c6f/attachment.htm>
Leandro Dardini
2012-Mar-25 09:15 UTC
[asterisk-users] Routing premature media to the calling channel
All NAT and firewall problems are already been excluded. All peers are on public IP address and no firewall is active between them. The missing routing of the audio path to the peer has been checked with tcpdump ... nothing is coming out from the asterisk box. Leandro 2012/3/25 Alex Balashov <abalashov at evaristesys.com>> I assume you have ruled out NAT and firewall issues? > > Between those two, 99% of the reasons why something may not be routed > somewhere correctly are accounted for. > > If you don't know, your best bet is to take a packet capture or SIP > debug on the Asterisk server and find out where that early media is going. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Atlanta, GA 30030 > Tel: +1-678-954-0671 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com > > > Leandro Dardini <ldardini at gmail.com> wrote: > > Hello, > I have a problem with premature media and inband progress audio. I am > using the latest 1.8.10.1 and this is the setup: > > soft phone --- asterisk --- SIP provider > > The number I call is giving back some hints via inband audio I am not able > to ear from the soft phone. They stop on the asterisk and are not routed > down the path to the sip phone. > > The SIP part is simple: > > soft phone -> asterisk: INVITE > > asterisk -> soft phone: TRYING > > asterisk -> provider: INVITE > > asterisk -> soft phone: 180 RINGING > > provider -> asterisk: 183 SESSION PROGRESS > > provider -> asterisk: AUDIO > > Unfortunately the AUDIO received from the provider by the asterisk box is > not sent to the soft phone. > > I think I have tried every combination of progressinband and > prematuremedia, without success. > > How can I made the audio received from the provider to the asterisk be > transmitted to the soft phone? > > Thank you > > Leandro > > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120325/f71ed692/attachment.htm>
Alex Balashov
2012-Mar-25 09:22 UTC
[asterisk-users] Routing premature media to the calling channel
Are you absolutely sure that nothing is coming out, even on a different interface than the one on which you are capturing? Are you capture on the Asterisk server and not the receiving host? Secondly, are you absolutely positive that something is supposed to be coming out? 183 does not logically imply or mandate backward early media, though 183+SDP is generally used as a convention to indicate that it is about to be sent. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini <ldardini at gmail.com> wrote:>All NAT and firewall problems are already been excluded. All peers are on >public IP address and no firewall is active between them. The missing >routing of the audio path to the peer has been checked with tcpdump ... >nothing is coming out from the asterisk box. > >Leandro > >2012/3/25 Alex Balashov <abalashov at evaristesys.com> > >> I assume you have ruled out NAT and firewall issues? >> >> Between those two, 99% of the reasons why something may not be routed >> somewhere correctly are accounted for. >> >> If you don't know, your best bet is to take a packet capture or SIP >> debug on the Asterisk server and find out where that early media is going. >> >> -- >> Alex Balashov - Principal >> Evariste Systems LLC >> 235 E Ponce de Leon Ave >> Suite 106 >> Atlanta, GA 30030 >> Tel: +1-678-954-0671 >> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >> >> >> Leandro Dardini <ldardini at gmail.com> wrote: >> >> Hello, >> I have a problem with premature media and inband progress audio. I am >> using the latest 1.8.10.1 and this is the setup: >> >> soft phone --- asterisk --- SIP provider >> >> The number I call is giving back some hints via inband audio I am not able >> to ear from the soft phone. They stop on the asterisk and are not routed >> down the path to the sip phone. >> >> The SIP part is simple: >> >> soft phone -> asterisk: INVITE >> >> asterisk -> soft phone: TRYING >> >> asterisk -> provider: INVITE >> >> asterisk -> soft phone: 180 RINGING >> >> provider -> asterisk: 183 SESSION PROGRESS >> >> provider -> asterisk: AUDIO >> >> Unfortunately the AUDIO received from the provider by the asterisk box is >> not sent to the soft phone. >> >> I think I have tried every combination of progressinband and >> prematuremedia, without success. >> >> How can I made the audio received from the provider to the asterisk be >> transmitted to the soft phone? >> >> Thank you >> >> Leandro >> >> >> >> -- >> _____________________________________________________________________ >> -- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120325/7b863487/attachment.htm>
Leandro Dardini
2012-Mar-25 09:35 UTC
[asterisk-users] Routing premature media to the calling channel
The asterisk box has only one interface. I am capturing all the traffic on the box and the only audio traffic is from the provider to the asterisk box. Obviously if I set progressinband=yes, then I get the ringing tone from the asterisk box, but no the audio from the provider I was looking for. Leandro 2012/3/25 Alex Balashov <abalashov at evaristesys.com>> Are you absolutely sure that nothing is coming out, even on a different > interface than the one on which you are capturing? Are you capture on the > Asterisk server and not the receiving host? > > Secondly, are you absolutely positive that something is supposed to be > coming out? 183 does not logically imply or mandate backward early media, > though 183+SDP is generally used as a convention to indicate that it is > about to be sent. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Atlanta, GA 30030 > Tel: +1-678-954-0671 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com > > Leandro Dardini <ldardini at gmail.com> wrote: > > All NAT and firewall problems are already been excluded. All peers are on > public IP address and no firewall is active between them. The missing > routing of the audio path to the peer has been checked with tcpdump ... > nothing is coming out from the asterisk box. > > Leandro > > 2012/3/25 Alex Balashov <abalashov at evaristesys.com> > >> I assume you have ruled out NAT and firewall issues? >> >> Between those two, 99% of the reasons why something may not be routed >> somewhere correctly are accounted for. >> >> If you don't know, your best bet is to take a packet capture or SIP >> debug on the Asterisk server and find out where that early media is going. >> >> -- >> Alex Balashov - Principal >> Evariste Systems LLC >> 235 E Ponce de Leon Ave >> Suite 106 >> Atlanta, GA 30030 >> Tel: +1-678-954-0671 >> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >> >> >> Leandro Dardini <ldardini at gmail.com> wrote: >> >> Hello, >> I have a problem with premature media and inband progress audio. I am >> using the latest 1.8.10.1 and this is the setup: >> >> soft phone --- asterisk --- SIP provider >> >> The number I call is giving back some hints via inband audio I am not >> able to ear from the soft phone. They stop on the asterisk and are not >> routed down the path to the sip phone. >> >> The SIP part is simple: >> >> soft phone -> asterisk: INVITE >> >> asterisk -> soft phone: TRYING >> >> asterisk -> provider: INVITE >> >> asterisk -> soft phone: 180 RINGING >> >> provider -> asterisk: 183 SESSION PROGRESS >> >> provider -> asterisk: AUDIO >> >> Unfortunately the AUDIO received from the provider by the asterisk box is >> not sent to the soft phone. >> >> I think I have tried every combination of progressinband and >> prematuremedia, without success. >> >> How can I made the audio received from the provider to the asterisk be >> transmitted to the soft phone? >> >> Thank you >> >> Leandro >> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120325/0cb35796/attachment.htm>
Alex Balashov
2012-Mar-25 09:37 UTC
[asterisk-users] Routing premature media to the calling channel
I think I may have misunderstood your initial question, sorry. You are looking for Asterisk to directly pass through the early media from upstream? Why would it do that? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini <ldardini at gmail.com> wrote:>The asterisk box has only one interface. I am capturing all the traffic on >the box and the only audio traffic is from the provider to the asterisk box. > >Obviously if I set progressinband=yes, then I get the ringing tone from the >asterisk box, but no the audio from the provider I was looking for. > >Leandro > >2012/3/25 Alex Balashov <abalashov at evaristesys.com> > >> Are you absolutely sure that nothing is coming out, even on a different >> interface than the one on which you are capturing? Are you capture on the >> Asterisk server and not the receiving host? >> >> Secondly, are you absolutely positive that something is supposed to be >> coming out? 183 does not logically imply or mandate backward early media, >> though 183+SDP is generally used as a convention to indicate that it is >> about to be sent. >> >> -- >> Alex Balashov - Principal >> Evariste Systems LLC >> 235 E Ponce de Leon Ave >> Suite 106 >> Atlanta, GA 30030 >> Tel: +1-678-954-0671 >> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >> >> Leandro Dardini <ldardini at gmail.com> wrote: >> >> All NAT and firewall problems are already been excluded. All peers are on >> public IP address and no firewall is active between them. The missing >> routing of the audio path to the peer has been checked with tcpdump ... >> nothing is coming out from the asterisk box. >> >> Leandro >> >> 2012/3/25 Alex Balashov <abalashov at evaristesys.com> >> >>> I assume you have ruled out NAT and firewall issues? >>> >>> Between those two, 99% of the reasons why something may not be routed >>> somewhere correctly are accounted for. >>> >>> If you don't know, your best bet is to take a packet capture or SIP >>> debug on the Asterisk server and find out where that early media is going. >>> >>> -- >>> Alex Balashov - Principal >>> Evariste Systems LLC >>> 235 E Ponce de Leon Ave >>> Suite 106 >>> Atlanta, GA 30030 >>> Tel: +1-678-954-0671 >>> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >>> >>> >>> Leandro Dardini <ldardini at gmail.com> wrote: >>> >>> Hello, >>> I have a problem with premature media and inband progress audio. I am >>> using the latest 1.8.10.1 and this is the setup: >>> >>> soft phone --- asterisk --- SIP provider >>> >>> The number I call is giving back some hints via inband audio I am not >>> able to ear from the soft phone. They stop on the asterisk and are not >>> routed down the path to the sip phone. >>> >>> The SIP part is simple: >>> >>> soft phone -> asterisk: INVITE >>> >>> asterisk -> soft phone: TRYING >>> >>> asterisk -> provider: INVITE >>> >>> asterisk -> soft phone: 180 RINGING >>> >>> provider -> asterisk: 183 SESSION PROGRESS >>> >>> provider -> asterisk: AUDIO >>> >>> Unfortunately the AUDIO received from the provider by the asterisk box is >>> not sent to the soft phone. >>> >>> I think I have tried every combination of progressinband and >>> prematuremedia, without success. >>> >>> How can I made the audio received from the provider to the asterisk be >>> transmitted to the soft phone? >>> >>> Thank you >>> >>> Leandro >>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >> > >-- >_____________________________________________________________________ >-- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120325/2f0dde7a/attachment.htm>
Leandro Dardini
2012-Mar-25 09:46 UTC
[asterisk-users] Routing premature media to the calling channel
I want to have the early media to pass from the provider down to the soft phone because it contains important information about the call, like "Your call cannot go through, please try your call again " ... The provider is giving this info via early media, just after the 183 SESSION PROGRESS. Leandro 2012/3/25 Alex Balashov <abalashov at evaristesys.com>> I think I may have misunderstood your initial question, sorry. > > You are looking for Asterisk to directly pass through the early media from > upstream? Why would it do that? > > > -- > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Atlanta, GA 30030 > Tel: +1-678-954-0671 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com > > Leandro Dardini <ldardini at gmail.com> wrote: > > The asterisk box has only one interface. I am capturing all the traffic on > the box and the only audio traffic is from the provider to the asterisk box. > > Obviously if I set progressinband=yes, then I get the ringing tone from > the asterisk box, but no the audio from the provider I was looking for. > > Leandro > > 2012/3/25 Alex Balashov <abalashov at evaristesys.com> > >> Are you absolutely sure that nothing is coming out, even on a different >> interface than the one on which you are capturing? Are you capture on the >> Asterisk server and not the receiving host? >> >> Secondly, are you absolutely positive that something is supposed to be >> coming out? 183 does not logically imply or mandate backward early media, >> though 183+SDP is generally used as a convention to indicate that it is >> about to be sent. >> >> -- >> Alex Balashov - Principal >> Evariste Systems LLC >> 235 E Ponce de Leon Ave >> Suite 106 >> Atlanta, GA 30030 >> Tel: +1-678-954-0671 >> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >> >> Leandro Dardini <ldardini at gmail.com> wrote: >> >> All NAT and firewall problems are already been excluded. All peers are on >> public IP address and no firewall is active between them. The missing >> routing of the audio path to the peer has been checked with tcpdump ... >> nothing is coming out from the asterisk box. >> >> Leandro >> >> 2012/3/25 Alex Balashov <abalashov at evaristesys.com> >> >>> I assume you have ruled out NAT and firewall issues? >>> >>> Between those two, 99% of the reasons why something may not be routed >>> somewhere correctly are accounted for. >>> >>> If you don't know, your best bet is to take a packet capture or SIP >>> debug on the Asterisk server and find out where that early media is going. >>> >>> -- >>> Alex Balashov - Principal >>> Evariste Systems LLC >>> 235 E Ponce de Leon Ave >>> Suite 106 >>> Atlanta, GA 30030 >>> Tel: +1-678-954-0671 >>> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >>> >>> >>> Leandro Dardini <ldardini at gmail.com> wrote: >>> >>> Hello, >>> I have a problem with premature media and inband progress audio. I am >>> using the latest 1.8.10.1 and this is the setup: >>> >>> soft phone --- asterisk --- SIP provider >>> >>> The number I call is giving back some hints via inband audio I am not >>> able to ear from the soft phone. They stop on the asterisk and are not >>> routed down the path to the sip phone. >>> >>> The SIP part is simple: >>> >>> soft phone -> asterisk: INVITE >>> >>> asterisk -> soft phone: TRYING >>> >>> asterisk -> provider: INVITE >>> >>> asterisk -> soft phone: 180 RINGING >>> >>> provider -> asterisk: 183 SESSION PROGRESS >>> >>> provider -> asterisk: AUDIO >>> >>> Unfortunately the AUDIO received from the provider by the asterisk box >>> is not sent to the soft phone. >>> >>> I think I have tried every combination of progressinband and >>> prematuremedia, without success. >>> >>> How can I made the audio received from the provider to the asterisk be >>> transmitted to the soft phone? >>> >>> Thank you >>> >>> Leandro >>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120325/ef09a48b/attachment.htm>
Alex Balashov
2012-Mar-25 09:51 UTC
[asterisk-users] Routing premature media to the calling channel
As far as I know, this is not the general tendency of any B2BUA that generates such media independently. However, I could be mistaken. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini <ldardini at gmail.com> wrote:>I want to have the early media to pass from the provider down to the soft >phone because it contains important information about the call, like "Your >call cannot go through, please try your call again " ... The provider is >giving this info via early media, just after the 183 SESSION PROGRESS. > >Leandro > >2012/3/25 Alex Balashov <abalashov at evaristesys.com> > >> I think I may have misunderstood your initial question, sorry. >> >> You are looking for Asterisk to directly pass through the early media from >> upstream? Why would it do that? >> >> >> -- >> Alex Balashov - Principal >> Evariste Systems LLC >> 235 E Ponce de Leon Ave >> Suite 106 >> Atlanta, GA 30030 >> Tel: +1-678-954-0671 >> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >> >> Leandro Dardini <ldardini at gmail.com> wrote: >> >> The asterisk box has only one interface. I am capturing all the traffic on >> the box and the only audio traffic is from the provider to the asterisk box. >> >> Obviously if I set progressinband=yes, then I get the ringing tone from >> the asterisk box, but no the audio from the provider I was looking for. >> >> Leandro >> >> 2012/3/25 Alex Balashov <abalashov at evaristesys.com> >> >>> Are you absolutely sure that nothing is coming out, even on a different >>> interface than the one on which you are capturing? Are you capture on the >>> Asterisk server and not the receiving host? >>> >>> Secondly, are you absolutely positive that something is supposed to be >>> coming out? 183 does not logically imply or mandate backward early media, >>> though 183+SDP is generally used as a convention to indicate that it is >>> about to be sent. >>> >>> -- >>> Alex Balashov - Principal >>> Evariste Systems LLC >>> 235 E Ponce de Leon Ave >>> Suite 106 >>> Atlanta, GA 30030 >>> Tel: +1-678-954-0671 >>> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >>> >>> Leandro Dardini <ldardini at gmail.com> wrote: >>> >>> All NAT and firewall problems are already been excluded. All peers are on >>> public IP address and no firewall is active between them. The missing >>> routing of the audio path to the peer has been checked with tcpdump ... >>> nothing is coming out from the asterisk box. >>> >>> Leandro >>> >>> 2012/3/25 Alex Balashov <abalashov at evaristesys.com> >>> >>>> I assume you have ruled out NAT and firewall issues? >>>> >>>> Between those two, 99% of the reasons why something may not be routed >>>> somewhere correctly are accounted for. >>>> >>>> If you don't know, your best bet is to take a packet capture or SIP >>>> debug on the Asterisk server and find out where that early media is going. >>>> >>>> -- >>>> Alex Balashov - Principal >>>> Evariste Systems LLC >>>> 235 E Ponce de Leon Ave >>>> Suite 106 >>>> Atlanta, GA 30030 >>>> Tel: +1-678-954-0671 >>>> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >>>> >>>> >>>> Leandro Dardini <ldardini at gmail.com> wrote: >>>> >>>> Hello, >>>> I have a problem with premature media and inband progress audio. I am >>>> using the latest 1.8.10.1 and this is the setup: >>>> >>>> soft phone --- asterisk --- SIP provider >>>> >>>> The number I call is giving back some hints via inband audio I am not >>>> able to ear from the soft phone. They stop on the asterisk and are not >>>> routed down the path to the sip phone. >>>> >>>> The SIP part is simple: >>>> >>>> soft phone -> asterisk: INVITE >>>> >>>> asterisk -> soft phone: TRYING >>>> >>>> asterisk -> provider: INVITE >>>> >>>> asterisk -> soft phone: 180 RINGING >>>> >>>> provider -> asterisk: 183 SESSION PROGRESS >>>> >>>> provider -> asterisk: AUDIO >>>> >>>> Unfortunately the AUDIO received from the provider by the asterisk box >>>> is not sent to the soft phone. >>>> >>>> I think I have tried every combination of progressinband and >>>> prematuremedia, without success. >>>> >>>> How can I made the audio received from the provider to the asterisk be >>>> transmitted to the soft phone? >>>> >>>> Thank you >>>> >>>> Leandro >>>> >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120325/922d8281/attachment.htm>
isrlgb at gmail.com
2012-Mar-25 13:44 UTC
[asterisk-users] Routing premature media to the calling channel
Do you have "r" in your dial string?
If yes remove that
-----Original Message-----
From: Leandro Dardini <ldardini at gmail.com>
Sender: asterisk-users-bounces at lists.digium.com
Date: Sun, 25 Mar 2012 11:35:45
To: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users at
lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Routing premature media to the calling channel
--
_____________________________________________________________________
-- 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
Leandro Dardini
2012-Mar-25 19:05 UTC
[asterisk-users] Routing premature media to the calling channel
Bingo, it was the "r" option! Thank you Leandro 2012/3/25 <isrlgb at gmail.com>> Do you have "r" in your dial string? > If yes remove that > -----Original Message----- > From: Leandro Dardini <ldardini at gmail.com> > Sender: asterisk-users-bounces at lists.digium.com > Date: Sun, 25 Mar 2012 11:35:45 > To: Asterisk Users Mailing List - Non-Commercial Discussion< > asterisk-users at lists.digium.com> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Subject: Re: [asterisk-users] Routing premature media to the calling > channel > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120325/e2754d98/attachment.htm>