similar to: Pressing number 2 in dialplan

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I want to detect brute-force password hacking attacks - thus if there are too many failed login attempts for a SIP account I want to "lock" this account. Does somebody have any ideas how this could be implemented? thanks klaus

Hello! What kind of sms text messaging capabilities does Asterisk have? I do not know very much about about SMS technology, but I am looking for the following features: 1. mobile SIP devices can send and receive SMS messages 2. Asterisk server be able to accept and send SMS messages through PRI lines and Internet connections. I noticed that Asterisk has an SMS function, but I am not farmiliar

Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a

Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have "rtpmap:18 g729" in their SDP, things work fine with Digium's commercial g729 license. How do I get "98 g729a" recognized by Asterisk? Thanks,

Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot

Hello! I am trying to configure my digium TE220 dual-span pci express card with Dahdi. I seemed to have managed to set up the card with the Dahdi kernel, as demonstrated by executing dahdi_scan: [1] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=1 totchans=31 irq=16

Hi, Does anyone have any experience with a Cisco 7961 + 7914 Operator Console setup and speeddials/BLF on Asterisk 1.4? Would appreciate feedback if this works reliably. I have a 7961 on skinny registering on an 1.4.19 box with chan_sccp and speeddials work fine so that part seems ok. I have no experience with the BLF part. >From googling around it seems that BLF on the 7914 only works with

I'm setting up a 7921 and now want to add a second line to the phone. In my SCCP.conf file I have: autologin = 235,299 However, on reloading SCCP the phone fails to login to the second line with this error: [Sep 12 12:46:49] WARNING[12224]: sccp_actions.c:185 sccp_handle_register: SEP001BD457F8B1: Failed to autolog into 299: Max available lines phone limit reached 299 Is there a

Hi, I need to subscribe and use several Polycom-Kirk 5020 handsets along non-Polycom-Kirk handsets on a one-cell Polycom-Kirk 600/3 base station. Has anyone tried this ? Which values did you pick for Subscription mode (with or without Account Code) and IPEI ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have two Asterisk servers running on the same LAN. One starts fine, but when I start the other I get: pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use and Asterisk does not start. OK I thought, I'll just change the port in dundi.conf. I changed it to 4521 and indeed it started just fine, until I reboot that is. WHen I reboot, I get the same

Hi list, After recently setting up voicemail for Asterisk 1.4.14 on my Debian etch server, I noticed that I can't delete any old voicemail messages. The voicemail menu option "Press 7 to delete this message" is available, but when I press 7 the response is always "message undeleted" and the message is still there. What could I be missing here? Thanks, Jaap

Hello, and please forgive the OT question. I'm just becoming desperate. I need two ISDN 30 circuits in Amsterdam, and I can't seem to be able to get a provider. I've tried KPN and Versatel. I'm based in California. Does anyone have any recommendations? Thanks in advance, Jose.

Friends, I will be attending FOSDEM in Brussells Feb 23-24. Anyone else? Me and Philippe Sultan (the Jabber/XMPP Asterisk developer) will be there, so we could have a SIP/XMPP/Asterisk ad hoc meeting :-) On Thursday, Feb 21, I will be in Utrecht, Netherlands for the free Open Telephony conference at Media Plaza. There's still seats available and a really good talk about ENUM with Patrik

Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Thanks in advance, regards, Rob.

Hi, I just upgraded from 1.2 to 1.4. In 1.2, when I did a "show uptime" I used to see a second line telling me the time since the last reload. Has this been removed in 1.4? The following is the output of my two test boxes: Connected to Asterisk 1.4.18.1 currently running on voip2 (pid = 10605) Verbosity is at least 3 voip2*CLI> show uptime System uptime: 15 hours, 55 seconds

Hello, everyone. Anybody know when that svn will be available again? Regards *Alex Montoanelli* Administra??o e Ger?ncia de Redes Unetvale Conectividade <http://www.unetvale.net> +55 48 3263 8700 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081126/cfe339fd/attachment.htm

starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these "timing" modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what

Is there a bullet type document with the features each version of Asterisk has? I know you can read the CHANGES file but that is not something you give a customer. I just need a one or two page document with bullet points showing the features added from 1.2 to 1.4 and from 1.4 to 1.6. Anyone know of an existing document or it this a make your own moment? -- Telecomunicaciones Abiertas de

This happens mysteriously & randomly. If asterisk was killed and restarted, it often gives this error myast*CLI> core stop now No such command 'core stop now' (type 'core show help core' for other possible commands) Any hint Thanks Jim