similar to: rtp.c dtmf issues solved.

http://bugs.digium.com/bug_view_page.php?bug_id=0001589 Has anyone else heard an audible blip, break or garble between answer and the native bridge attempt using sip? If I change the usleep(500000); to usleep(5000); in rtp.c the proble totally goes away... even the note above it says it needs to be fixed. bkw -------------- next part -------------- An HTML attachment was scrubbed... URL:

http://bugs.digium.com/bug_view_page.php?bug_id=0002945 Test it.. I couldn't sleep tonight... thought I would see if I could find and fix it... Also did this gem too for ya... http://bugs.digium.com/bug_view_page.php?bug_id=0002948 bkw

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

http://bugs.digium.com/bug_view_page.php?bug_id=0001019 "This patch allows to bind RTP flows to a specific interface, additionally the SDP session descriptor get's coherent with the same address that is used for RTP traffic, this includes sip<->sip and sip<->voicemail and others(not tested, but should work). Maybe some problems with NAT appear, if anyone notices any bug

Hi everyone, I've just finished the 'SWait' app for *. SWait = Super Wait :) Syntax: SWait([timeout][cim]) 'timeout' is the number of seconds to wait. Defaults to 'ResponseTimeout'. 'c' is for continue. This changes the default behavior of the app from performing a 'Goto(t,1)' on timeout, to a 'Goto(pri+1)' on timeout, ie. continue to

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

Hello ! My asterisk log is full of messages like this: [2011-03-06 19:01:15] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:20] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:25] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06

Users cannot dial 0 to get to the operator in voicemail. * 1.0.9 Linux asterisk1.local 2.4.21-32.EL #1 Wed May 18 18:31:54 EDT 2005 i686 athlon i386 GNU/Linux (CentOS) Snom 360 DTMF=RFC2833 Switched LAN, no problems w/ DTMF anywhere operator=yes in voicemail.conf this does not apply to my situation: http://bugs.digium.com/bug_view_page.php?bug_id=0003080 When OGM is played, you press 0,

Are there any known problems converting dtmf from oob over iax2 to inband over rtp/ulaw? Obviously it works when converting to inband over pri/ulaw et al, but how about rtp? I've got packet traces that confirm that 2833 packets are properly generated when I have 2833 configured for the rtp link, but the other side seems to be ignoring those packets. So I tried inband on that link; nothing

Hi Can asterisk support inband dtmf? I have an sip device which doesn't support out of band dtmf Sean

Dear friends, As I commented some while ago in the list, occasionally when DTMF Tones are sent, they appear in RTP Payload and in Events too, producing duplicate tones being recognized. This behavior happens in Asterisk as well as in Gateways such as Cisco, for which we had the opportunity to observe the error and extensively debug it. We ended up recognizing good digits by adjusting audio gain

A friend and I have recently added the ability to announce the callers position in the call queue every x seconds.. or even just inject an anouncement every x seconds. All setup in queues.conf and can be setup per queue. My next project is to add the ability to announce the callers estimated wait time. I want some feedback to see whats the best method to calculate that? What do you want just

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem,

Hi there, I have serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542

So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=0000889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the

Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Should this patch work against current cvs? Of the 3 files 2 are .patch and one is .diff - what's the difference between them, and how should I

Hello, When using Asterisk version 0.7.2, FreeBSD port with Mediatrix 1124 gateway, there is problem with DTMF "out-of-band". See debug below: Mediatrix forces (*) to use Payload Type as 96: [...] a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [...] Then we've got this nice debug from (*): May

I'm playing with codecs at the moment and have found some notices errors when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was getting garbled sound, but after changing magic number for both codecs to 97 (as per http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT,

Hi Everyone! I live in Sweden and can not get CallerID to work on analog incoming lines. I m trying to find out if DTMF style CallerID works on a FXO card (X100). I`v seen one solution with a modem attached in parallel with the X100 just to provide the ID on its serial port. It must be much better if this can be implemented in to the X100 driver. Any info about this would be highly appreciated.

Can anybody recommend a good web interface for asterisk that actually works. I am looking for a web interface that can show how many callers are on the phone, should be able to transfer the calls and disconnect. I have tried using the flash operator but has been unsuccessful in making it work. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com