similar to: Long Delay after sip reload command

Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the

I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. I recognize that some action might be required by another provider which is outside Vitelity's control, but it seems that they should at least be trying to help resolve the problem by helping me determine

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 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

Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John

Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them ......... and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are

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

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

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.

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