similar to: crashes after upgrade from 1.2.16 to 1.4.21.2

Hi, Using asterisk 1.4.21.2. For some calls (usally telemarketers) entering through a BRI zap channel I somtimes notice the callerid on my polycom 601 phone and the CDR's 'src' field don't match. They are even totally different. And the displayed callerid is nowhere to be seen in the CDR record. Is there a rational explanation? -- http://www.lesculturelles.net

Hi, I've got a problem with my asterisk set up which has been going on for a while (months). I'm currently running 1.2.7.1 on a gentoo box with the topology below: +-----+ PSTN ---------+ * +------------- Service Provider (wctdm400p) +-+-+-+ IAX | | | | FXS --+ +-- SIP (cisco 7940)

Hi, When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Is there a way to avoid these warnings? Or at least turn them off when at the

Hello im trying to play some videos with totem on CentOS 8. I get the following message in the command line. "totem ** Message: 17:22:06.269: Missing plugin: gstreamer|1.0|totem|MPEG-1 Video decoder|decoder-video/mpeg, mpegversion=(int)1, systemstream=(boolean)false (MPEG-1 Video decoder) (totem:12230): Gtk-WARNING **: 17:22:06.270: Drawing a gadget with negative dimensions. Did you forget

Hi, I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by "zap show channels". I tried adding "dahdichanname = no" to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel

Hi >i have this data > X [1] 5.79 1579.52 2323.70 68.85 426.07 110.29 108.29 1067.60 17.05 22.66 [11] 21.02 175.88 139.07 144.12 20.46 43.40 194.90 47.30 7.74 0.40 [21] 82.85 9.88 89.29 215.10 1.75 0.79 15.93 3.91 0.27 0.69 [31] 100.58 27.80 13.95 53.24 0.96 4.15 0.19 0.78 8.01 31.75 [41] 7.35 6.50

flight 12230 xen-unstable real [real] http://www.chiark.greenend.org.uk/~xensrcts/logs/12230/ Regressions :-( Tests which did not succeed and are blocking, including tests which could not be run: test-amd64-i386-xl 9 guest-start fail REGR. vs. 12228 Regressions which are regarded as allowable (not blocking): test-amd64-i386-qemuu-rhel6hvm-amd 7 redhat-install

Hi, Ii try to connect an Asterisk server running 1.4.21.2 version with gtalk2voip services. Everything is fine till the call for DTMF test: there is no audio and Asterisk shows [Nov 18 14:51:47] WARNING[20502]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission SIPCALL-435578583-1984100284 at 72.20.112.114 for seqno 1 (Critical Response) [Nov 18 14:51:47] WARNING[20502]:

Hi guys.. I just wanted to know if this config could work correctly, since a lot of u guys have been working a lot with asterisk... Asterisk 1.4.21.2 -> chosen because is the last with Zaptel support ? Zaptel 1.4.12 Add ons 1.4.9 Currently i have a 1.6.1.0 Server running with no problems, but I'm about to buy a openvox a400p Card, so i would like to test both environments, 1.4 and 1.6,

Hi list, I just experienced an odd behaviour in 1.4.22 vs 1.4.21.2. To cut a long story short, IAX2 is not tx-ing hangup... Scenario is composed of two asterisk systems A and B. A receives calls from IAX users X, Y, Z, etc, does some validation and forwards them to B, also over IAX. When B hangs up, it transmits IAX hangup which A receives who, in turn, does not transmit the IAX hangup to its

The Asterisk.org development team has released Asterisk versions 1.4.21.2 and 1.2.30. Both of these releases include fixes for two security issues. Both of these issues affect users of the IAX2 channel driver. For more details on these vulnerabilities, see the published security advisories, AST-2008-010 and AST-2008-011. AST-2008-010: Asterisk IAX 'POKE' resource exhaustion -

The Asterisk.org development team has released Asterisk versions 1.4.21.2 and 1.2.30. Both of these releases include fixes for two security issues. Both of these issues affect users of the IAX2 channel driver. For more details on these vulnerabilities, see the published security advisories, AST-2008-010 and AST-2008-011. AST-2008-010: Asterisk IAX 'POKE' resource exhaustion -

Hi, We are experiencing this issue of redial dtmf tones generated randomly in the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one FXS is used for Fax and rest are empy) connected to the netgear switch and all the phones are connected to this switch and there are no non sip devices in the

Hi, all. I've been searching google, bug reports and forums and have looked in all the asterisk-users list archives back to 2003 but haven't seen an answer to this, so thought I'd post here. The problem seems to be that Asterisk 1.6.0.5 is sending backslashes (needed to escape commas and so forth in 1.4.21.2) as *literal* backslashes to Mysql, so that Mysql gives a syntax error

Hello I'm using the Read() function to play a message prompting for the user to type a number followed by the # key to validate, with a 30s time-out and 2 tries: ============== [test] exten => s,1,Wait(2) exten => s,n,Answer ;typed DTMF: prompt for number to dial: 2 tries, 30s time-tout exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,,,2,30) exten =>

Hi all! My problem is that calls being placed in the queue, and are waiting while the agents are busy, when an agents is then free they gets connected to the agent but there is silence (no voice). If a caller has not to wait in the queue, there is no problem. My agents have an iax2 client, and imcoming calls are over SIP. queue.conf: persistentmembers=yes autofill=yes ringinuse=no

Ok so I'm trying to install some games from big fish, just to see if they'll work. I download the game and when I run it it tells me: "You don't have permission to install games from Big Fish Games on this computer. Please log in with an administrator account and try again. How do I fix this?

Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: ---------------------------------- [CC] res_adsi.c -> res_adsi.o [LD] res_adsi.o -> res_adsi.so [CC] res_agi.c -> res_agi.o [LD] res_agi.o -> res_agi.so [CC] res_clioriginate.c -> res_clioriginate.o

On attempting to make Zaptel 1.2.16 on FC5, I get the following messages: /usr/src/zaptel-1.2.16/xpp/xbus-core.c: In function 'debugfs_open': /usr/src/zaptel-1.2.16/xpp/xbus-core.c:171: error: 'struct inode' has no member named 'u' make[3]: *** [/usr/src/zaptel-1.2.16/xpp/xbus-core.o] Error 1 make[2]: *** [/usr/src/zaptel-1.2.16/xpp] Error 2 make[1]: ***

Hi, I am facing the following issues with call parking: 1) Asterisk does not announce the parking extension:- I press # and hear "transfer" followed by a beep. I dial 700 but do not hear the parking extension number. 2) After parking time, the original extension rings and i accept the call. When I again try to park, the call park sequence fails. I have attached features.conf and