similar to: asterisk 1.4.21.2 a caller waited in queue, after connect to agent hears silence

Hi Everyone, We use the top buttons on Aastra 55i to login and logout from Queues. This is the order: Button 1 = Login to English Queue Button 2 = Login to Spanish Queue Button 3 = Logout of English/Spanish Queues There are indicator LEDs on each of these buttons. Is there anyway we can send a SIP request or some other communication to get the Aastra 6755i phone to keep the LED for login set

CVS HEAD from about 1 week ago. TDM30P and call through Nufone. I was talking and wanted to park the call and move to another phone to pick it up. I hit #701 instead of #700 though -- after a pause, I got a fast busy and the call was gone. When I called the person back, she said that Alison told HER that 701 was an invalid extension. I should have heard that though, not her. If I dial

Hi, the subject says almost everything. For a better understandig, here is the background: after sending pathworks servers to EOL we installed a OS/2 server system. And we waited a long time to go to the next evolutionary step. Until now. We are using file and print services. The user count is growing to 150. We have 2 locations connected by a VPN. Our own wish has been a NAS with

Hi, As mentioned we have the problem that sometimes (could be up to a view times a day) for the calling party (SIP Device) you here ringing. The called party however answered the phone, but hears nothing. The calling party keeps ringing until the phone is hangup. First I thought maybe the card or the server has a problem, so I changed from a PCI beronet 4bri to a Junghanns 4bri PCIexpress and

First of all - Many, many thanks to Mark for his troubleshooting and fix of bug 1320 (FXO_KS signalled Zap Channels on Adtran 750 Channel Bank Stuck in Rsrvd State). I have heard complaints that once every couple weeks, when a user picks up their analog phone (t1 span off of a TE410P into an Adtran 750 with 6 FXS cards), they don't get dialtone, but instead, hear another conversation.

https://bugzilla.netfilter.org/show_bug.cgi?id=1728 Bug ID: 1728 Summary: Regression: iptables lock is now waited for without --wait Product: iptables Version: 1.8.x Hardware: x86_64 OS: All Status: NEW Severity: normal Priority: P5 Component: unknown

Hallo, das Subject sagt schon fast alles. Zum besseren Verst?ndnis der Hintergrund. nachdem ich vor vielen jahren Pathworks-Server durch ein OS/2-Serversystem abgel?st habe,haben wir es auch noch geschafft, die n?chste Evolutionsstufe bis jetzt herauszuziehen. Wir nutzen File- und Printservices. Das ist alles. Die Anzahl der User tendiert gegen 150. Wir sind an zwei Standorten, aber ?ber VPN

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, After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we experience crashes at random intervals with: [Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame read(0, <unfinished ...> +++ killed by SIGSEGV (core dumped) +++ Process 15755 detached On a second sister-machine with a mirror install we have the same problem. So it doesn't seem to be a

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, 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

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, 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

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

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

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

Hello there! I've problem with setting up unicall / mfcR2. can't find proper notation for channel, trying unicall/1, unicall/1/1001, unicall/g1, unicall/g1/1000 and still having no luck. klaudia*CLI> !cp call /var/spool/asterisk/outgoing -- Attempting call on Unicall/1001 for application Dial(363) (Retry 1) Mar 25 09:29:34 NOTICE[19920]: channel.c:2429 __ast_request_and_dial:

We have a network with Windows NT/2000, SGI's, Linux, Macintosh, AIX. Our user account/passwords for all our machines is handled by NIS on an AIX machine. Our unix machines autheticate through NIS. Our windows machines authenticate by loging on to an NT server acting as the PDC, and that PDC has a service installed on it called "WinDD NIS" (made by Tektronix I think) which contacts