Massimo Nuvoli
2010-Jul-07 08:13 UTC
[asterisk-users] This may be a problem.. Answer not working on 1.4.32 over SIP trunk..
I found a strange thing on Asterisk 1.4.32, the same defect on 1.4.26.? I spend 4 hours to try to solve... but found only a workaround. As is easy to reproduce the problem i need to know if this is a bug or if there is some idiot configuration that i miss. Maybe also the bug is know... Scenario: Asterisk installation on ubuntu 9.04 64 bit. Trunk SIP (two different providers) On the Asterisk server there are a number of SIP clients. If i use the sip client all things ok, i made a call and everything ok. If i place the call from the server (or if i call trhu the SIP trunk the asterisk system) everytime the Answer() application seeems to NOT work. The only way to make it work is to use some other function that do the Answer in place. (call coming from the SIP trunk) If i use Answer() MusicOnHold() I hear nothing. If i use Answer() PlayBack(silence/1) MusicOnHold() or Answer() VoiceMail(1234 at default) i can hear all ok (it seems that the PlayBack and the VoiceMail apps are able to Answer really...) I checked the SIP debug trace, it seems no problem on the SIP side. Thnks guys. -------------- next part -------------- A non-text attachment was scrubbed... Name: massimo.vcf Type: text/x-vcard Size: 354 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100707/5be2c152/attachment.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 250 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100707/5be2c152/attachment.pgp
Zeeshan Zakaria
2010-Jul-07 14:22 UTC
[asterisk-users] This may be a problem.. Answer not working on 1.4.32 over SIP trunk..
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-07 4:16 AM, "Massimo Nuvoli" <massimo at archivio.it>
wrote:
I found a strange thing on Asterisk 1.4.32, the same defect on 1.4.26.?
I spend 4 hours to try to solve... but found only a workaround.
As is easy to reproduce the problem i need to know if this is a bug or
if there is some idiot configuration that i miss.
Maybe also the bug is know...
Scenario:
Asterisk installation on ubuntu 9.04 64 bit.
Trunk SIP (two different providers)
On the Asterisk server there are a number of SIP clients.
If i use the sip client all things ok, i made a call and everything ok.
If i place the call from the server (or if i call trhu the SIP trunk
the asterisk system) everytime the Answer() application seeems to NOT
work.
The only way to make it work is to use some other function that do the
Answer in place.
(call coming from the SIP trunk)
If i use
Answer()
MusicOnHold()
I hear nothing.
If i use
Answer()
PlayBack(silence/1)
MusicOnHold()
or
Answer()
VoiceMail(1234 at default)
i can hear all ok (it seems that the PlayBack and the VoiceMail apps
are able to Answer really...)
I checked the SIP debug trace, it seems no problem on the SIP side.
Thnks guys.
--
_____________________________________________________________________
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Zeeshan Zakaria
2010-Jul-07 14:25 UTC
[asterisk-users] This may be a problem.. Answer not working on 1.4.32 over SIP trunk..
I have two test asterisk boxes, both version 1.4.26, on which I do Answer()
followed by MusicOnHold() and it works just fine. I do this all the time as
this is my standard way of testing new contexts.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-07 4:16 AM, "Massimo Nuvoli" <massimo at archivio.it>
wrote:
I found a strange thing on Asterisk 1.4.32, the same defect on 1.4.26.?
I spend 4 hours to try to solve... but found only a workaround.
As is easy to reproduce the problem i need to know if this is a bug or
if there is some idiot configuration that i miss.
Maybe also the bug is know...
Scenario:
Asterisk installation on ubuntu 9.04 64 bit.
Trunk SIP (two different providers)
On the Asterisk server there are a number of SIP clients.
If i use the sip client all things ok, i made a call and everything ok.
If i place the call from the server (or if i call trhu the SIP trunk
the asterisk system) everytime the Answer() application seeems to NOT
work.
The only way to make it work is to use some other function that do the
Answer in place.
(call coming from the SIP trunk)
If i use
Answer()
MusicOnHold()
I hear nothing.
If i use
Answer()
PlayBack(silence/1)
MusicOnHold()
or
Answer()
VoiceMail(1234 at default)
i can hear all ok (it seems that the PlayBack and the VoiceMail apps
are able to Answer really...)
I checked the SIP debug trace, it seems no problem on the SIP side.
Thnks guys.
--
_____________________________________________________________________
-- 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
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Massimo Nuvoli
2010-Jul-09 06:10 UTC
[asterisk-users] This may be a problem.. Answer not working on 1.4.32 over SIP trunk..
Zeeshan Zakaria ha scritto:> I have two test asterisk boxes, both version 1.4.26, on which I do > Answer() followed by MusicOnHold() and it works just fine. I do this all > the time as this is my standard way of testing new contexts.Yesterday i tested another installation and i found the same issue. Maybe the problem is "SIP" related or "console channel" related. I explain (if someone can do a test i am happy). Go to the asterisk console, place a "dial" command calling thru the SIP trunk, then place a "transfer" to the extension MusicOnHold after the Answer... (this is the sequence) dial 0number at from-sip (the from-sip is the context where a sip phone can dial to the trunk) pick up the phone called transfer *199 at from-sip (the *199 extension is "Answer -> MusicOnHold") you must hear the music on the phone called (or not) So this may be a "console channel problem"... Yesterday i try to use the outgoing spool (place a file on /var/spool/asterisk/outgoing making a call to the phone and directly go to the *199 extension, the same thing i do on console automated with no console channel), audio ok. So i am going to open a bug... :-) Thnks.> Zeeshan A Zakaria > > -- > www.ilovetovoip.com <http://www.ilovetovoip.com> > >> On 2010-07-07 4:16 AM, "Massimo Nuvoli" <massimo at archivio.it >> <mailto:massimo at archivio.it>> wrote: >> >> I found a strange thing on Asterisk 1.4.32, the same defect on 1.4.26.? >> >> I spend 4 hours to try to solve... but found only a workaround. >> >> As is easy to reproduce the problem i need to know if this is a bug or >> if there is some idiot configuration that i miss. >> >> Maybe also the bug is know... >> >> Scenario: >> >> Asterisk installation on ubuntu 9.04 64 bit. >> >> Trunk SIP (two different providers) >> >> On the Asterisk server there are a number of SIP clients. >> >> If i use the sip client all things ok, i made a call and everything ok. >> >> If i place the call from the server (or if i call trhu the SIP trunk >> the asterisk system) everytime the Answer() application seeems to NOT >> work. >> >> The only way to make it work is to use some other function that do the >> Answer in place. >> >> (call coming from the SIP trunk) >> If i use >> >> Answer() >> MusicOnHold() >> >> I hear nothing. >> >> If i use >> >> Answer() >> PlayBack(silence/1) >> MusicOnHold() >> >> or >> >> Answer() >> VoiceMail(1234 at default) >> >> i can hear all ok (it seems that the PlayBack and the VoiceMail apps >> are able to Answer really...) >> >> I checked the SIP debug trace, it seems no problem on the SIP side. >> >> Thnks guys. >> >> -- >> _____________________________________________________________________ >> -- 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 -------------- A non-text attachment was scrubbed... Name: massimo.vcf Type: text/x-vcard Size: 354 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100709/7cdb8fca/attachment.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 250 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100709/7cdb8fca/attachment.pgp