similar to: Ring Busy ?

Hi i am not a expert on Asterisk and search a lot of small information : I use Asterisk 1.6.1.4 with MySQL. That's work and in my extension.conf, i have: [as5300-incoming] switch => Realtime and in extconfig.conf extensions => mysql,general,VOIP_Extensions A lot of Extension are into the table VOIP_Extensions. I am search to know if i can add a :

Hi actually, i test a new Asterisk Server and i want add Mysql Realtime SIP. I read on the wiki: =================================================== Database Config put the following in res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = myuser dbpass = mypass dbport = 3306 Values in sip.conf or iax.conf like in older versions of * are no longer used. Database Table Lets

Hi, I ma using Asterisk 1.2.18 & FreePBX 2.2.1. I have assigned every users in office with Polycom with 2 extensions as below 555 8555 I have configured Follow-me to ring when the users doesn't picks the phone on line 1(555) after 10 seconds & then ring the line 2(8555). But this is not a standard telephony which I have been advised to change like below. If someone calls

Hi, I have an asterisk box connected to a PRI LINE, some extensions are trunked by IAX to another box that's connected via ISDN BRI to a PBX. That's what's happening call comes in via PRI to the first box and is sent to the other box exten => _N.,1,Dial(IAX2/sip:pippo@trunk/*${EXTEN}) the other box rings a pbx (simulating an ISDN call from a BRI line) exten =>

Hello, I am looking to know if it is possible to modify the SQL query that is on Realtime sip accounts. I want multiple servers use the same sql table, so getting an extra "server" field to indicate that the data is valid on the X server is this possible? thank you in advance jerome

I know that with Voicemail you can either do voicemail(u<extension>) or voicemail(b<extension>), but with the Sipura SPA-841's I need to be able to roll lines from one extension to an alternate on the phone. For example: If extension 100 is busy, it will ring extension 120 on the same phone, and if that is busy it will ring 140, and then if it hits line 4 with no response, it will

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

Hello, Has anybody seen that Audiocodes gateway is replying with "486 Busy here" when it's actually not (last call ended ~15 seconds ago). I see this quite often. From other logs i see that previous call ends at 11:13:01, then app_queue tries to dial at 11:13:14 and fails numerous times, before succeeding at 11:14:02 I have attached sample SIP debug log: Any ideas what i could

I have a ring group set up with 3 extensions. we'll use 14, 15 and 16. When a call comes in, it rings all three extensions. If one particular extension already is on the phone, it completely skips that phone and only rings the other 2. Example to explanation sake is: Call comes in, ext. 14 is already in the middle of a call, 15 and 16 will ring normally, but 14 does not have any

Hi we have a Samba-Winbind 3.5.10 and i have a big quantity of errors in logs : [2012/09/24 16:01:29.248286, 1] winbindd/winbindd_ads.c:728(lookup_usergroups_memberof) lookup_usergroups_memberof ads_search member=CN=TESTDD,OU=Villeurbanne,OU=ExEA,DC=ELIOTT,DC=fr: Out of memory [2012/09/24 16:03:38.498533, 1] libads/ldap_utils.c:323(ads_ranged_search_internal) could not pull first

Hi! I am having difficultly in having users of various SIP devices obtain the correct behaviour when they call a busy number ie. only hearing the Congestion/Busy tone. I assume this might be because the SIP device itself generates the 'ring' tone? With my current setup in the dialplan extract (below) the user of the SIP device hears one 'ring' and then the busy tone if a number

Hello, I'm using asterisk 1.6.0.1 and realtime queues. But when I make changes in database (for example: change strategy from ringall to random), but asterisk shows old strategy, doesn't update this parameter. My question is, how I can dynamically change ring strategy. Thanks in advance. -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML

Hi, I have written an click2dial application that rings an agent soft phone and connects the agent with a customer. very often I can see, that the agent softphones signal a busy back to server, although the phone is definitely hung up and the previous calls where handled normally. I testet 3cx Version 6 and X-Lite V1 up to V3... all show the same misbehaviour. I did a SIP Trace and can see,

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

Hi anyone know this error: Mar 20 12:01:06 gw winbindd[14756]: [2009/03/20 12:01:06, 0] sam/idmap_tdb.c:db_allocate_id(106) Mar 20 12:01:06 gw winbindd[14756]: idmap Fatal Error: UID range full!! (max: 20000) Mar 20 12:01:06 gw winbindd[14756]: [2009/03/20 12:01:06, 0] sam/idmap_tdb.c:db_allocate_id(106) Mar 20 12:01:06 gw winbindd[14756]: idmap Fatal Error: UID range full!! (max: 20000)

Hi A question, i have upgraded a beta serveur from Asterisk 1.6.1 to 1.6.2 and now all SIP Relatime user are rejeted : [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21639 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21658 handle_incoming: Ignoring SIP message because of retransmit (REGISTER Seqno 44199, ours 44199) [Oct 5

When agents are on the phone, and the CLI queue show command shows their status as busy, the queue still tries to send them calls. Running Asterisk 1.6.0.17 and using AddQueueMember to dynamically add agents. ringinuse is set to no for queue. Agents are using Polycom 430s. dialplan: exten => s,n,Queue(orders,itT,,,80) queue definition in queues.conf: [orders] maxlen=20 queue-thankyou=

Hi all, I have 12 SIP phones at a particular site all connected to a local asterisk server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming calls. An IAX gateway is used for outbound calls. At the moment, when an incoming call comes in, asterisk dials every SIP phone like so: Dial (SIP/1&SIP/2&etc.) This has worked fine for some months, but I noticed a few days ago

Running AsteriskNow, with FreePBX front end. Have two users who use softphones on notebooks in the field. Problem is that if the softphone receives a call, but the user is not available to pick it up, Asterisk will send the call to voice mail as normal, but the softphone will continue to ring...... I've tested the Virbiage and x-Lite softphones, same issue. Is there an IAX setting in

Hi I have two small question, if you can help me ;=) Problems with Answer+Music my extension: [Cal-In] exten => _811XXXX20,1,Goto(C-Internal,100,1) exten => _811XXXX21,1,Goto(C-Internal,200,1) [C-Phibee] exten => 100,1,Ringing exten => 100,2,Wait,1 exten => 100,3,Answer exten => 100,4,Dial(SIP/201&SIP/200,30) exten => 100,5,Hangup exten =>