similar to: Asterisk 1.6.1.12 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.6.0.20. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.20 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * clarify requirecalltoken option in iax.sample.conf

The Asterisk Development Team has announced the release of Asterisk 1.6.0.20. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.20 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * clarify requirecalltoken option in iax.sample.conf

Hi All, I'm in the lab with Asterisk 1.6.1.12 and several ATA's testing T38. I hit a snag with the Grandstream HT502. It only seems to nail up a session at 9600bps. The Grandstream GXW4104 nails up consistently at 14400bps. I'm using the same equipment in the same configuration, just switching out the ATA. I have the latest firmware on each unit. Any ideas on what could cause

Hi All, I've been knee deep in T38 faxing for a couple of weeks now, trying to find a version of Asterisk that would pass through T38 with an Audiocodes Mediant 1000 and MP203 ATA. I had problems with 1.6.0.x through 1.6.1.10. Tested 6 different versions. Either it just would not work or fail back to G.711, or re-invite with wrong T38FaxMaxDatagram sizes, faxes would work one-way and not

With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} is [un]set in an odd way. for example consider: 999,1,Swift(some long message that you dont want to wait for|5000|5) 999,n,NoOp(DTMF: ${SWIFT_DTMF}) if while I am listening to the playback, i interrupt and dial: - "12345", SWIFT_DTMF is set to

Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come across is that the peer is congested and sending RTCP messages asking us to slow the RTP down. Is there any way we can verify this? [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412121

Hello everybody, When I execute the "sip show peers" command in the asterisk console I always get the following notice, repeated 15 times after the sip show peers output. [Dec 21 03:38:31] NOTICE[12693]: utils.c:1074 ast_wait_for_output: Timed out trying to write This happens on a freshly installed 1.6.1.12 and a 1.6.1.6 box that I am running. Both of them use Debian Linux (lenny) on

I'm playing around with an ALIX 2D2 board (http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an AMD Geode processor with 256MB of RAM. Also available are two network interfaces, two USB ports and one serial port (no keyboard or VGA). I'm using the Voyage Linux distro, which basically is Debian Lenny optimized for this board. Asterisk 1.6.1.12 runs fine on the

Relatively new to Asterisk and SIP and am trying to run a proof of concept using Asterisk to make an outbound call through an Audiocodes gateway via SIP using Asterisk version 1.6.1.12. The specific requirements of the gateway in the configuration I am trying to use specify that the Name part of the From header be blank with the outbound number that needs to be dialed in the number field of

Hi All, Maybe someone knows this, im using dahdi in combination with a TDM400, where 2 analog PSTN lines are connected. The weird thing is tho that when someone calls the analog lines it goes perfectly fine, the line comes in and all works ok. Except: Sometimes the callerid from the caller is not the complete number, but only a few random numbers from that phonenumber, and sometimes its complete.

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries:

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries:

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

When I place an outbound call from asterisk 1.6.1.12 to a FXO port on my Cisco 1760V 12.4, the channel changes - seemingly incrementing: e.g., in the first call, below, the channel name is "SIP/vgw1-00000075" -- the second call (on the same FXO port after a soft hangup on the CLI) is "SIP/vgw1-00000077" How can I extract this information in the dialplan so that I can use

Hi everybody, I would like to use realtime authentification with my LDAP. My Asterisk is v. 1.6.1.12. I'm using OpenLDAP The command realtime ldap status is OK. I have configure these files : /etc/asterisk/extconfig /etc/asterisk/res_ldap.conf /etc/asterisk/extensions.ael I do nothing and I have that in my console : [Jan 21 10:41:48] WARNING[30020]: res_config_ldap.c:809

The Asterisk Development Team has announced the release of Asterisk 1.4.28. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.28 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Send ack (response/message) after receiving manager action

The Asterisk Development Team has announced the release of Asterisk 1.4.28. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.28 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Send ack (response/message) after receiving manager action

Hello, I have Asterisk 1.6.1.12 with FAX For Asterisk Components: Applications: 1.6.1.5_1.1.6 Digium FAX Driver: 1.6.1.5_1.1.6 (optimized for core2_32) If I use call file with spool -------------------------------------------- Channel: SIP/IP/DEst No MaxRetries: 0 RetryTime: 10 WaitTime: 50 Application:SendFAX Data:/var/spool/asterisk/test.tif

Hello, We are trying to send faxes by T.38 protocol to a remote SIP proxy from a local extension. The local extension sends the INVITE, Asterisk sends the call to the Proxy the call is connected with a regular audio codec. After a few seconds the remote proxy sends an INVITE with UDPTL and the Asterisk sends it to the local extension and it's accepted, but (here the problem starts) just

I'm trying the following command on the source machine: virsh migrate --live --copy-storage-all --verbose TEST qemu+ssh:// 10.30.76.66/system If I ssh into the destination machine when this command is running, I can see NBD copying data as expected, and if I wait long enough it completes and succeeds. However if I ctrl+c this command above before it completes, it causes virsh commands