similar to: Asterisk 1.6.2.16 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.4.39. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.39 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 1.4.39. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.39 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 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 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 release:

The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 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 release:

Hello, I keep receiving errors such as the following: Error rebooting domain: Timed out during operation: cannot acquire state change lock (held by qemuProcessReconnect) Traceback (most recent call last): File "/usr/share/virt-manager/virtManager/asyncjob.py", line 90, in cb_wrapper callback(asyncjob, *args, **kwargs) File

Hi every one, When i run asterisk i am getting error as ERROR[2109]: chan_sip.c:3963 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data ............ i am unable to understand what it is and how come.......... can any one help me regarding this issue.......... Hope i get a positive reply...... With Regards, viswavardhan --------------

I have a very simple setup exactly as described in the HOWTO section " 4.2. Routing for multiple uplinks/providers". One is cable (eth1: dhcp) and the other is PPPoE (ppp0). I used the following commands to configure the routing once all of my interfaces are up and i have configured SNATing for them: ip route add 66.11.173.0/24 dev ppp0 src 66.11.173.224 table 11 ip route add default

Hi, My phones stopped auto-answering when being paged, since I moved on to Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk 1.6.2.16. I looked at the wiki but nothing I try there works, even if I cut and paste the same setup. Any one has any idea of what I should change from my old 3.2.3 setup? My older phone (501) still using 3.1.6 still auto-answer

Adding some links to links.html --- images/hw/pioneer_c.png | Bin 0 -> 3389 bytes images/hw/tesla.png | Bin 0 -> 6145 bytes links.html | 13 +++++++++++++ 3 files changed, 13 insertions(+) create mode 100644 images/hw/pioneer_c.png create mode 100644 images/hw/tesla.png diff --git a/images/hw/pioneer_c.png b/images/hw/pioneer_c.png new file mode 100644 index

Hi, I've been playing with agent penalties (Asterisk 1.6.2.16). I've set up multiple agents, and as documented agents with the lowest penalty value get called first. The issue I have is that if the agent with the very lowest penalty does not answer, instead of going with to the second best agent the queue rings the same agent over and over and over again. I've realized that

After installing msxml6_x64.msi for 64 bit configured wine, when my 64bit windows executable gets started, am getting the following error: There was already a bug raised with number 18126 on this. Please share any ideas how to resolve the same. ................................................................................................. err:ole:COMPOBJ_DllList_Add couldn't load in-process

Hi, is there a way to know if originate call channel ended the call *before* connecting to context/extension/priority? DIALSTATUS is empty, HANGUPCAUSE is always 16, nothing in SIP Headers nor in AST_CONTROL_FRAME_[HANGUP|ANSWER] Asterisk is 1.6.2.16 Thanks for any hint -- Daniel

Hi, in asterisk 1.6.2.16 I get spurious DTMF recognition over SIP from an Audiocodes. I think the DTMF recognition is the Audiocdes' fault, the Audiocodes log seems to say so as well, but I want to make sure, and fixing the Audiocodes is not an option in this particular case - don't ask. Can someone explain to me what the following means *exactly* [Feb 10 21:15:40] DTMF[2538] channel.c:

This issue is peripherally related to the following thread Re: [Dovecot] client certs with godaddy ssl cert This is running on CentOS 5.2 with latest Atrpms for Dovecot as of this weekend. # rpm -qa | grep dovecot dovecot-sieve-1.1.5-8.el5 dovecot-1.1.4-0_81.el5 With assistance from Rainer Frey (Inxmail GmbH), I am able to successfully use client ssl certs for imap access on both my Nokia

Hello, Our Asterisk runs with multiple remote sites (12 over an MPLS network), everything works fine except for the last site we have juste installed. When VOIP flows comes/goes from/to this site, there are sound quality issues, persistent, 100% reproducible, on every call. This is not a bandwidth or latency or jitter problem, everything is fine on the network. Our MPLS provider does all check

Hi, We had 2 asterisk 1.4 connected together in iax, all was fine. One of them was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38 When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But calling from 1.6.2 to 1.4 give a bad audio to calling party (words are cutted, you can't understand the words). On callee party it's still good. We replace

Hi everyone, I have been trying to get T.38 Faxing to work with Iristel sip trunks for last few days but havn't been sccussful. I am using Asterisk 1.6.2.8 and SpanDSP 0.6. Here is what I see in the tcpdump capture: 1. Call come in from the trunk as regular voice call with g.711 codec 2. Asterisk answers the call and recognizes the CNG and sends the call to fax extension 3. Eventually

Alex, Thank you so much for your response. I've been so consumed with other business that I only just now getting back to this issue. We have implemented your suggestion which is perfect. Thank you again. I've never asked a question of the community before and I'm extremely happy with the rapid response I received. Somewhat related to this initial problem I have an additional

I am having a problem with an Asterisk 1.6.2.15 server that runs a small call center with Queuemetrics. In the past month we've had this problem 3 times. The problem is that Asterisk simply stops responding. No calls in or out and you cannot even get to the CLI. The process seems to be running but there is simple no activity. All I see in the log files is: [Jan 14 16:30:46]

Hello list. I'm trying to rebuild the 2.6.9.67.0.20.EL kernel, but it fails even without modifications. How did I try it? Created a (non-root) build environment (not a mock ) Installed the kernel.scr.rpm and did a rpmbuild -ba --target=`uname -m` kernel-2.6.spec 2> prep-err.log | tee prep-out.log The build failed at the end: Processing files: kernel-xenU-devel-2.6.9-67.0.20.EL Checking