search for: nachum

Hi everyone, I am running asterisk with release 12.1.0.rc3 and PJSIP. I have a peer which sends OPTIONS method for session keep-alive, and the asterisk is not responding to it. That of course disconnects the call after a few minutes. Is there a settings in the PJSIP.conf to respond to in dialog OPTIONS method? Looking at the documentation I haven't seen it. Does anybody know a workaround?

Hello Asterisk users, We noticed that on Asterisk 12 zombie processes are being generated - They are released after a while, but we have around 10-20 zombie processes running. We are not sure if this is a normal behavior or an issue. We saw in the documentation that the bridging module creates zombie processes - is it related? Thank you, Yaron. -------------- next part -------------- An HTML

Hello all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF: 1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF settings according to the incoming INVITE - RFC2833 or inband. The is no such settings in PJSIP. Do you know is there is a plan to develop it? 2. When we setup 2 peers, one RFC4733 and the other inband,

Hello everyone, I am trying to activate Music On Hold using DB on Asterisk 13. It works fine but in order to use new Music On hold definitions I have to reload the moh module. - The following is my configuration in extconfig.conf - I added the following line: musiconhold.conf => mysql,asterisk,bit_ast_config - The following is the table in the database: mysql> select * from

Hello Asterisk Users, I have been looking for similar auto dtmf mode implementation on pjsip, but didn't see it coming, so I decided to give it a try. My basic plan was to do it as simple as possible with minimum changes because I am not familiar with all Asterisk code. My idea is to use rfc4733 settings, but when going over the codecs check if telephone-event appear and if not set the dtmf

Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? Below you can see the output of the asterisk monitor. <--- Received SIP request

Ok Mark Michelson. Thank you very much! You answer tells me that I was in the wrong path trying to access information from SIP 183 message. I need to find a way to let the callee pass information/data to the caller, even before accepting the call. That is, send data during the ringing time. And in my case, there will be more than one callee ringing at same time. As ASTERISK will not forward each

On 2/23/15 3:03 AM, Yaron Nachum wrote: > Hello everyone, > I am trying to activate Music On Hold using DB on Asterisk 13. > It works fine but in order to use new Music On hold definitions I have > to reload the moh module. > > - The following is my configuration in extconfig.conf - I added the > following l...

Hello everyone, I have started testing the PJSIP stack. I saw that it is possible to setup statically multiple AOR contacts, setup qualify_timeout and attach it to an endpoint, and then dial using this endpoint. When I setup the configuration I used the cli in order to see the status of the contacts, and it worked fine - whenever a contact is unreachable, the status is updated to Unavailable.

Hi everyone, I am starting to work with PJSIP on release 12.1.0.rc3. I used to have Asterisk 1.8 with the regular sip channel. I was using the usereqphone settings in order to set user=phone on from and to URIs. Is there a similar config in PJSIP? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Asterisk users and developers, The last few weeks we had several crashes on live asterisks running versions 12.2.0rc1 / 12.6.1 with PJPROJECT versions 2.1.0 / 2.2.1. We opened a ticket - ASTERISK-24471. After investigating the issue I can say that the scenario is a CANCEL being received while handling ANSWER and before generating the 200OK response. Looking at the core file we see that

Hi, I am planning to use Dovecot 2.0 as a Mail server for a Voicemail system. There is not much traffic so I am not concerend about scaling. However redudancy is an issue. I need to have two Dovecot servers installed on two different farms and setup HA between the two. I would prefer that the HA would be automatically. I have two different storage machines with SAN, and I can setup Layer 2 between

Hello everyone, I am upgrading from release 1.8 and I have a strange behavior with CDR generation. We are using a Redirect server for Number portability, and I see that once the call is going through the Redirect Server additional CDR records are generated - we have 3 additional records. This Behavior is different then what we had on Release 1.8. Does anyone have a clue how to remove these CDR

Hello, I have an issue with Asterisk 12 PJSIP. When receving an INVITE with FROM URI that has a port number, the Asterisk removes the port from URI on consecutive Responses / Requests. This causes an issue with one of our SIP servers (it doesn't recognize the response / request). Below you can see an incoming INVITE and the outgoing 200OK response. I have highlighted the issue in Yellow. Does