Last week I migrated some of our servers to Asterisk 1.4.18 and we started
seeing audio drops of several seconds during SIP calls. After investigating
it we noticed that Asterisk was increasing the RTP timestamps abnormally
during a conversation.
I'm including a text file with a subset of the
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know for sure what's going on here? I could go back to using Monitor,
I suppose, but MixMonitor is somewhat less hacky.
Thanks
jurgen
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We're using a Cisco 3660 as a T.38 gateway for fax reception, but as
Asterisk does not support T.38 in pass through mode yet what we're doing
is sending a SIP REFER message (via the Transfer application) to our SIP
provider (when we detect fax tones) to redirect the call to the Cisco
gateway.
, I didn't tried yet.
How do you send that SIP REFER to the Cisco?
Gracias,
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose
Comellas
Sent: Thursday, June 15, 2006 9:02 AM
To: Asterisk Users Mailing List - Non-Commercial
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.
Thanks.
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using the
G.711u (ulaw) codec between both machines inside a LAN.
On Tue November 15 2005 23:19, George Vagenas wrote:
Juan Jose Comellas wrote:
Has anybody ever used the TxFAX application to send a fax to RxFAX on
another Asterisk installation. I'm trying to do just that and both apps
remain
is that both apps block on a call to ast_waitfor() with a
inifinite timeout. I've seen this in several other places in Asterisk and
these calls are normally the source of hung channels. Is this correct?
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free to send every kind of disappointments opinions. That is
going to feel me much better that no answers.
(Even if you can show me how stupid I was doing all kind of mistakes)
Regards,
Carlos Alperin
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.I can hear the
other person but they can't hear me. Has anyone had this before?
Regards,
Chris
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0.6.6pre3-4
libopenh323-1.15.3c21.15.3-4
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Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose
Comellas
Sent: Friday, September 30, 2005 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial
I want to start experimenting with Asterisk running on a router with embedded
Linux using the OpenWRT firmware. Has anybody tried routers other than the
Linksys WRT54G or WRT54GS for this purpose? What do you recommend?
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Just in case somebody else has this problem, it seems that there is a bug in
the 3.1.3a version of firmware of the Sipura SPA-841. Updating to the 3.1.4a
version of the firmware solved the problem.
On Sun August 28 2005 01:55, Juan Jose Comellas wrote:
I have just bought several Sipura SPA
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by a firmware
change!
John Novack
Juan Jose Comellas wrote:
I have just bought several Sipura SPA-841 SIP phones, and after some
testing I have found out that the volume received by other parties when
calling using the handset is very low. I've been able to reproduce this
problem in the 3
/05, Juan Jose Comellas [EMAIL PROTECTED] wrote:
I have just bought several Sipura SPA-841 SIP phones, and after some
testing I have found out that the volume received by other parties when
calling using the handset is very low. I've been able to reproduce this
problem in the 3 phones I've
configuration options
but nothing I has helped so far.
Has anybody else experienced this problem? There are only two holes for the
microphone in the handset and they are really small. I was thinking that
myabe this is the cause. Any thoughts?
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626-814-2354
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context=pstn-inbound-voice
channel = 2-7
Thanks
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the OpenLDAP server and Microsoft Active
Directory 2000.
You can download it from:
http://comellas.com.ar/app_ldap/app_ldap-0.1.0.tar.bz2
You'll need some knowledge of LDAP to be able to use this application
successfully.
Please report any problems you may have with it.
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This is a little bit off-topic, so forgive me. Does anybody know of any VoIP
phone line provider in Switzerland that supports a VoIP protocol that can be
connected with Asterisk?
Thanks.
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Jose Comellas
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the channels that are
inserted into the conference? The channels will mainly use SIP (maybe IAX2
too occasionally).
Thanks for your help.
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a
conversation.
A)Is it possible to * recognize this DTMF digits during the
conversation?
B) Can a softphone recognize this DTMF digits during the conversation?
C) How difficult it is?
Thanks,
Gerald
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it always in
memory?
Thanks.
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