similar to: Variables error in 1.8.6.0.

How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070803/c6d473ce/attachment.htm

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am not sure if its just me, but i am able to get only local channel variables containing RTCP QOS values. The Version is 1.8.14. I want to store values of bridged channel in CDR. Phone is Cisco 7941 SIP and with sip show channelstats i see all the relevant information (jitter,packet loss) i want to get. It even calculates packet loss in %. But i am not able to store it to CDR. Asterisk 1.4

Dear Tariq; About elastix.org, this can be use with Asterisk or it is coming as a complete IP Telephony, Call Center, IVR and Reporting? Because, I do not need to install another IP Telephony on the server which already has asterisk which is an IP Telephony, this will cause a problem in the service (for example, when listening for SIP port of 5060).

I found a clue as to why the second leg is not returning a local or remote address: [2023-07-06 11:40:35] WARNING[253072]: pjsip/dialplan_functions.c:903 channel_read_pjsip: No transport information for channel PJSIP/222-0000007d [2023-07-06 11:40:35] WARNING[935126]: func_channel.c:527 func_channel_read: Unknown or unavailable item requested: 'pjsip,local_addr' [2023-07-06 11:40:35]

Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve

Hello Asterisk users around the world! Recently, I have been working with pretty large Asterisk installations. 300 servers running Asterisk and Kamailio (OpenSER). Replacing large Nortel systems with just a few tiny boxes and other interesting solutions. Testing has been a large part of these projects. How much can we put into one Asterisk box? Calls per euro invested matters. So far,

I'm confused about something.... In Asterisk 1.4 you can collect RTP QoS metrics at the end of a call with: ${CHANNEL(rtpqos,audio,all)} Now, when your using the AMI to do a callout, like this... ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/1000 Variable: callid=849120 Variable: destination=SIP/1001 Variable: timeout=70000 Variable: timeout_warning=60000

Ok so now I'm getting this when doing a make in asterisk... travis at pcimphone1:~/downloads/asterisk-13.5.0$ make [LD] chan_pjsip.o pjsip/dialplan_functions.o -> chan_pjsip.so /usr/bin/ld: /usr/local/lib/libpjsip-ua-x86_64-unknown-linux-gnu.a(sip_inv.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC

Ok that did it after I did the steps to completely remove everything and do a new install. Thanks! > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Joshua Colp > Sent: Wednesday, September 23, 2015 10:12 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject:

Hello Group, We have dovecot deployed as solely a Pop3 service that is used by our applications to pass mail from one application to another internally. We have roughly 4 applications that connect to the Pop3 service every 2 seconds, to check for new messages and pop them for processing if they are present. Depending on the site, we have between 1024-2048MB of memory set for default_vsz_limit.

Hello all, I have two Tablegen questions in the context of an unconventional architecture. The pertinent details: the architecture has multiple register files that are selected via a bit in the instruction. One of the register files is a traditional one. The other is unconventional in that all source and dest registers are both read from and written to by the instructions. So add special0,

Hello, How can I go to the Leg B channel in Asterisk Dialplan _after_ call ends? I can use Dial G option to go to Leb B channel when call is answered, but how to go here when call ends? Is here any option/function in Dial Plan? Or should I use ast_bridged_channel(chan) to get bridged channel and try to retrieve data I need from internal structures using custom c module and Asterisk API?

The Asterisk Development Team has announced the release of Asterisk 1.8.27.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.27.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 1.8.27.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.27.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Mates, I'm trying to solve some log errors I getting. I have samba 3.0.7 installed on Suse 9.0. It has been happening intermittently for several months. I don't know if this is XP client related or what. Any thoughts: The errors are: Feb 9 20:00:33 skyline smbd[14408]: [2005/02/09 20:00:33, 0] lib/util_sock.c:get_peer_addr(1000) Feb 9 20:00:33 skyline smbd[14408]: getpeername

The Asterisk Development Team has announced the release of Asterisk 11.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

I have used rsync on many systems, and never had a problem. I am stumped on what to do with this. I have a box behind a LinkSys router and I can not "push" or "pull" data to or from it from anywhere. When I try, it logs into the remote server, gets the file list and just stops. The remote server shows a process running until I control-c out of it.when I do that, I get: rsync

The Asterisk Development Team has announced the release of Asterisk 12.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New