similar to: RTP Timeouts not clearing calls

Hi John This works fine providing extensions 1001,1002 and 1003 are "Incall" or "Paused" - the problem appears to be that is a handset say 1002 is "ringing" then the 2xxx then the penalty is not honoured. This is well described in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to

Hi Dovid Yes was one of the options but as the required list is dynamic becomes very messy - and all combinations problem - where as "call all workers job xxx" is what is needed so the ability to call 20+ numbers is what is needed - agi does a database search for all jobx workers and constructs a dialstring with SIP, DAHDI and Local devices. Can someone tell me where to find maximum

Hi All I have been looking at this problem for a few days/weeks now and after some advice please. I currently have a customer on 11.25.3 and I am in the process of upgrading versions and OS (Debian) and all things that involves mysql -> PDO etc The problem I have is the customer want a simple call distribution like this Extn 1001, 1002, 1003 to be called on an incoming call - if they

Paddy, Why not use local extensions? You can do something like this. Exten => s,1,Dial(Local/set1 at call_all&Local/set2 at call_all &Local/set3 at call_all) [call_all] Exten => set1,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105 Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111 Exten =>

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

As an example....if we have a call that: 1. originates via PSTN line to one of our local DID's in Seattle 2. comes into our Asterisk server in Los Angeles or Denver 3. is routed by Asterisk for termination back to a different Seattle PSTN ....and if our VOIP call termination provider requires (in order to get their best rate) all calls to go through their Nextone

Thank`s all, Chris Ziomkowski wrote: > If you only want to use H.323 with Asterisk, you should configure it as an H.323 gateway. > Why are you trying to set "softswitch"? I was asked by a costumer, because he could not use a asterisk as a softswitch in the Nextone configuration, so I`m looking for the difference in asterisk configs files. Thank`s a lot for you help...

Hello All, "safe_asterisk" just sent me an email saying "Asterisk on bill exited on signal 11. Might want to take a peek.". Looking at the /var/log/asterisk/message doesn't show me anything... This is a fresh installed Asterisk 1.6.2.6 on Ubuntu 9.10 (64-bit) and it is routing calls from Nextone MSW Softswitch to VPS Softswitch... Any reason why Asterisk died?

Paddy, It's pretty easy to spot from the CLI. A voicemail gets called. And the screen basically stops scrolling from there. Eventually you'll get the "Task processors exceeded 500 queued tasks" or something like that. And maybe channels attempting to hangup due to lack of RTP (If you have no-rtp timers configured). Once you find the problem mailbox, You can call it via any

Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...

Hi list I have a requirement that I just don't know how to address - I don't think its strange but can't find any pointers anywhere. I have a user that wishes to have a "multi phone" divert. By that I mean "calls made to his extension say Ext200 can be redirected to a different extension say Ext400 and also to his home landline. Doing the dial is fine using

Hi All I am trying to get voicemail switched over to ARA on version 13 and notice that the password is not stored in the db when it is changed. A little hair pulling and playing around and I think the problem is in the function ast_update2_realtime in main/config.c. Issued source is ==> int ast_update2_realtime(const char *family, ...) { RAII_VAR(struct ast_variable *,

Hello All, Got one customer and he is getting disconnection within 15 seconds when he tries to make outbound calls. Initially, it was working fine without any glitches... Other customers on the same system are working fine, its just with this customer only. This is the error message thrown by Asterisk on the CLI: - Jan 18 12:23:30 WARNING[30532]: chan_sip.c:1228 retrans_pkt: Maximum retries

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U)

Hi List I am having trouble running the command siptest:~# asterisk -rx 'dialplan reload' most times it does what I expect and I get a response as below siptest:~# asterisk -rx 'dialplan reload' Dialplan reloaded. every now and then I get no response i.e. siptest:~# asterisk -rx 'dialplan reload' siptest:~# and a "verbose 10" setting shows [Mar

Hello, we're in process of testing an interconnection with a trans-european carrier. But the carrier wants the SIP signalling to server 1 and the RTP stream to server 2. How do I configure asterisk to work with that type of installation. It seems they are using NexTone as SIP Signaling and RTP servers. Can someone help me??? Regards, Marc -- CTO Marc Storck

Hi Paddy! This used to be 80 characters total (including all characters like channel type, '&' and '/'. Had the same issue in the past where I extended that in the code and recompiled. From what I understand there is basically no longer a hard limit in Dial since the recent change in the latest versions other than a single device must not exceed this but you can concatenate

Hello all, Just wondering on a behavior I noticed while testing with realtime sip peers with names like 111.222 at mydomain.com. Using Kamailio as outbound proxy, it sends Asterisk a sip message where To header value is < sip:111.222 at mydomain.com> and From header has value "username" < sip:111.333 at mydomain.com;transport=UDP>;tag=fc609171. When Asterisk sends out the

Hi all A quick question about busy lamps I have lamps working 'sort-of' on my GXP2000 lamps flash with ringing and go solid red when call gets answered but stay green when a call is made from the extension. Setup is Ext 200, 201, 202, each monitor the other two when 200 calls 202 - the BLF on 200 and 201 flash red - when 202 answers both 200 and 201 show BLF for 200 as red but

Hi List I have had a look at the various posts on this and seem to be more confused than ever - but then again that's not hard ;-) I am using Version 1.4.33.1 build from the Debian "lenny" distros I am trying to implement a simple screening [macro-screen] exten => s,1,Background(press1) exten => s,n,WaitExten(5) exten => 1,1,NoOp(accepted) ; Dont set a reply so dial