similar to: No subject

'module show like sip' Cheers Nick.

Hopefully someone here can save my sanity. I have been trying to solve this problem for days now, but just cant put my finger on it. Im new to * so I have probably done something stupid! I have a TDM400P with one FXO module and a FXS module. The main problem I have is not being able to get the extension attached to the FXS module to ring or be able to make calls. It gets a dialtone fine but I

Hi guys. I'm trying to disable all debug output, but am not having any success: nick@asterisk-dev1:~> sudo asterisk -r Asterisk 1.2.8, Copyright (C) 1999 - 2006 Digium, Inc. and others. <..snip...> certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.8 currently running on

Afternoon All, Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to get a working SCCP channel). During the process Music On Hold appears to have died (Not, just when calling from a SCCP device, but coming in on SIP also). CLI is showing -- Executing [XXXXXXXX at unauthed-inbound:2] MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack

one under my default context at extention.conf. And what is [pbx_config]? Thanks Eyal -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, June 25, 2010 4:05 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Is there a default dial plan that is not in

DNS failure could do this Asterisk used to get stuck in a symmetric DNS request wait state which meant everything ground to a halt as it waited for a reply while DNS timed out. The recommended option was either ip only or a DNS proxy that failed fast this letting asterisk continue Cheers Duncan > On 27/05/2015, at 11:55 pm, Stefan Viljoen <viljoens at verishare.co.za> wrote: >

Hello, I have a Sipura SPA-3000 connected to my PSTN line and forwarding calls to my Asterisk box. It is a SIP peer "pstn-spa3k". I have setup "call-limit=1" in the peer config. When a call comes into Asterisk I get the correct "inuse" values but the hint isn't updated: sprite*CLI> sip show inuse * User name In use Limit * Peer name

When creating users in the sip.conf file, they do not appear when running the "sip show users" command from the CLI until i restart. A reload doesnt make them appear. As i said, I am new to the whole Asterisk thing, however have worked with IP/SIP PBX's for a few years - its most likely a user problem though! Check it out and let me know what you get. Cheers Chris PS - I would try

What kinds of problems were you having? I'm on 1.4.0 and chan_gtalk.so simply doesn't load. Of the 146 files in the /usr/lib/asterisk/modules/ directory, asterisk loads 144 of them, omitting only chan_gtalk.so and res_jabber.so. Connected to Asterisk 1.4.1 currently running on monkey (pid = 9371) Verbosity is at least 3 foo*CLI> module load chan_gtalk.so [Mar 7 10:23:07]

Recent posts indicate people have been having luck with the nokia E60/E7x phones and asterisk. I was wondering though if anyone had had any luck with the N80? I've got the N80 to register with asterisk, and that works just fine. However, it gives a 486 when I try to place SIP calls to it (either to the register username, or to the phone number). Oh, and I can't figure out how to make

-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, May 25, 2004 5:30 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #3891 - 8 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or

And after reload ALL your phones are unreachable for 2 minutes! Imagine you have several thousands devices unreachable for 2 minutes. How much calls will fail during that time? Regards, Mindaugas Kezys Kolmisoft UAB=20 VoIP Billing Solutions e-mail: info at kolmisoft.com URL: http://www.kolmisoft.com -----Original Message----- From: asterisk-users-bounces at lists.digium.com =

Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With SIP 3.2.X firmware (available on the Polycom download site) and Asterisk 1.6.1, Polycom phones now support a full featured BLF showing statuses of Ringing, Inuse and Online and one touch directed call pickup. On the asterisk side all that needs to be done is to add a hint to the extension and enable directed pickup.

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

Hello We have recently upgraded to Realtime engine (sip buddies and extensions) and now have problems with calling local SIP users. I have rtcachefriends=yes but tried with 'no' and it's even worse. (asterisk 1.8.1.1 + realtime mysql) Here's an example: User 1000 registers successfully and can then be called with Dial(SIP/1000,30) successfully After some time when I try to call

Hi, all I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database "mydatabase") via cdr_adaptive_odbc.

Hi, I am trying to integrate Asterisk with a very old PABX I have here for test purposes. I have it linked with and FXO module. Now the test scenario I am building goes like this: Incoming call on Legacy PABX --> Call Transferred to Asterisk --> Announcement Played --> Call Transferred to SIP Xtn --> If call is unanswered perform a hook flash on active zap channel and return it to

This week starts with the exciting news: We're getting close to Asterisk 1.0 again. After the failed attempt earlier this year, we've been able to remove a lot of the MAJOR/CRASH bugs from the bug tracker and Mark feel's it's time to target 1.0 again. At this point, the community needs to work as a community, spending extra time on finding bugs, solving issues, improving

use depending on what the subnet mask is. The output provided shows two possible networks: 172.31.253.0/24 and 172.31.254.0/24. Or is this all part of the same address space with a different mask? If it is all the same space, then is the asterisk server network stack properly configured with a proper subnet mask? The bb can reach the asterisk server because it registers. Hope this helps On

Hello using asterisk 1.8, compiled res_fax.so and res_fax_spandsp.so - both loaded successfuly in sip.conf set t38pt_udptl=yes but faxes still don't work even in passthru mode. if i do a 'sip show channel' on the channel via which i am sending fax it shows: T.38 support Yes however if i do sip show channel of my channel (from other server) it shows T.38 support