similar to: Play volume/speed adjustment on the per call basis

Hi Steve and all, 1. Faxing from asterisk back to the same asterisk (from one Zap channel to another) doesn't work for us. Txfax called with the 'caller' parameter issues CED, while the receiving side needs CNG in order to switch to fax extension with rxfax. 2. This is probably the reason why J2 and our UC don't recognize incoming fax. Thank you. Alex Zarubin Webley Systems

Hello, The v1-0_stable cvs release doesn't include the recent change ('poll' instead of 'select') in channel.c. Will it end up there any time soon, or we need to use cvs head to pick up this change? Thank you. Alex Zarubin Webley Systems -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, 1-2 sec pauses happen while * plays (streams) messages/prompts. We get reports about that from users and experience it ourselves randomly. Cannot reproduce it for debugging though, so need to figure out some other ways to fix it. 1. It's not silence recorded within or pauses between audio files 2. It's not load related - can happen with no load at all 3. We use decent boxes - dual

With multiple inbound PRIs (and hunting across them) coming to multiple [asterisk] servers it is important to be able to do administration, i.e. control which PRIs in the same hunt group take (and which don't take) calls from telco at any given period of time. Our pre-asterisk platform uses SERVICE commands for this purpose to put B-channels into 'out-of-service'/'maintenance'

Hi We are having a significant (> 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes. All the other legs are PSTN (TE410P). The example configuration Slave box 1 meetme <--- IAX2 ---> Master box meetme <--- IAX2 ---> Slave box 2 meetme The delay is between Slave box 1 and Slave box 2 The primary suspect is our iax configuration

Please, try exten => 99,1,Wait,1 exten => 99,2,Record,/tmp/pcmfile:pcm exten => 99,3,Wait,1 exten => 99,4,Playback,/tmp/pcmfile exten => 99,5,Wait,1 exten => 99,6,Record,/tmp/voxfile:vox exten => 99,7,Wait,1 exten => 99,8,Playback,/tmp/voxfile (put your own extension). Pcm recording is OK, playback is OK. Adpcm recording is noticeably worse. Adpcm playback is very

Hi, We have no problems sending to HP and Panasonic fax machines in the office. We do have problems when we try to send faxes to services supporting fax, i.e. J2 or our UC platform. The receiving side doesn't recognize fax. To send a fax we drop into /var/spool/asterisk/outgoing: Channel: Zap/g1/<fax number> MaxRetries: 0 WaitTime: 20 Context: webley_txfax Extension: txfax_ext

Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e "Error" -e "eventually" p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got

Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get this codec? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030916/b8be2453/attachment.htm

I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD

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The Asterisk Development Team has announced the release of Asterisk 1.8.24.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.24.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.8.24.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.24.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

If there are multiple NICs in the box, how do we specify the local IP address to be used for RTP? Anything in rtp.conf ? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030609/d151f190/attachment.htm

Can T400P be configured for nfas (one d-channel providing signaling for more than one span)? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030522/f2b637a9/attachment.htm

Hello, If I understand it correctly, when sending INVITE, o= and c= sdp fields are built using p->ourip IP address. At this point RTP packets will be coming to the default asterisk IP address. For the machine with multiple interfaces this could be not the right one (not what we want). Could it be configured (in rtp.conf or in sip.conf per context) ? Thank you. Alex Zarubin --------------

Hey all - Time for my silly question of the day, and before I bust out vi and dtrace... If there a simple, existing way I can observe the read / write / IOPS on a per-zvol basis? If not, is there interest in having one? Cheers! Nathan.

Current ISelDagToDag is created once per module. The TargetLowering class is allocated there and register classes are added and the computeRegisterProperties is called. In order to switch back and forth between mips16 and mips32, I need to be able to reset what is done during computerRegisterProperties. Has anyone else looked into this for another port? Ideas? Mips16 is an instruction

I'm implementing this capability to allow switching between mips32 and mips16 code generation on a per function basis (should be useful for arm to thumb switching too). The problem is that while various things are done on a per function basis, there are two passes registered on a per module basis (target lowering and instruction selection). With the new attribute scheme, we can wake up

http://bugzilla.mindrot.org/show_bug.cgi?id=210 ------- Additional Comments From markus at openbsd.org 2002-04-08 22:17 ------- check this http://www.mindrot.org/~djm/ssh-keynote/ ------- You are receiving this mail because: ------- You are the assignee for the bug, or are watching the assignee.