similar to: asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

Hi to all, I have a strange behavior in my asterisk server. I have a queue for 5 agents, the calls enter the queue an go to the agents normally, but if I need to transfer or dial directly to an agent extension that is already in a call, the pbx hung up the actual call (not the transferred call). This is what I see in the log. Called 103 -- Agent/103 is ringing --

I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Here is sip debug from pressing "transfer->blind->dial dest->Dial Key" (note both

I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a

On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > > Do you have any idea where to start to look at? Adding additional output > in the source code? Which functions could be interesting? I may add own > debug code to see why things are happening as they happen here. The logic for T.38 negotiation lives all in the res_pjsip_t38 module and the request to negotiate works using a

On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote: <snip> > > t38modem and asterisk are using > > m=image 35622 udptl t38 > ^^^^^ > > Provider uses > > m=image 35622 UDPTL t38 > ^^^^^ > > Could this be a problem? If I'm sending internal only, it's always > lowercase. Looking at the tests we have we

On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote: > Has anybody any idea why asterisk drops the media stream in the 200 OK? > The channel has been T38_ENABLED before! Or is it necessary to add more > debug code? Who does the negotiating? > Only asterisk or is pjsip doing some parts, too? Asterisk does the T.38 negotiation and produces the answer SDP, PJSIP does the SDP

Hello everyone, I've had the time to examine some specific serial/parallel forking scenarios with Asterisk lately. Looking at chan_sip it appears that anytime Asterisk wants to tear down a call before it's brought up, it sends a 603 Declined: } else { /* Incoming call, not up */ const char *res;

Hi, I have some problem with my provider. While the sip registration is successful, i intermittently encounter problem in dialing out. I receive 603 Declined error in my Sjphone client. The asterisk log shows line is busy/congestion. Appreciate if help or direction can be provided. Thanks. CK -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Sir, Please find attached the log file that I took from the asterik server during a call...Please check the SIP packets exchanged between OpenSer that send an Invite SIP packet to theasterisk server and the asterisk Server and let me know wat this DECLINEd message means... Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 05/12/2017 at 08:49 PM, Joshua Colp wrote: > On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote: > > <snip> > >> >> If I'm doing exactly the same call originated with another extension, >> there can't be seen these frequent changes. But the strange thing is, >> that in both cases the part between extension and asterisk doesn't show

The PJSIP endpoint is configured for ulaw only. Not sure how or why we are seeing the g729 on calls for this endpoint. Would this be a case that asterisk detects the rtp stream is g729 even though it's negotiated as ulaw? Why would asterisk change the format to g729 when disallow = all and allow = ulaw are the endpoint settings? [121] type = endpoint context = IS transport = transport1 aors

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Hello Everyone. In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to other routes if the chosen route rejects the call. Now, My current scenario is if I get "BUSY" back from the first provider, I send a busy back to my customer. If I get something like CHANUNAVAIL (Like a SIP 503) I advance to the next carrier and attempt the call. This works

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

In my dialplan I'd like to send a "603 Declined" message to the user placing the call. I see the commands for the Busy and Congestion, but not the one for the Declined. Any help? Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130726/5ac93551/attachment.htm>

Oh god it works ! to switch cisco to upd I used config: <transportLayerProtocol>2</transportLayerProtocol> with udp it works well, thanks for your help :) > On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote: > > If you use UDP with force_rport=no it'll work. > If you use TCP then set rewrite_contact=yes so it'll reuse the established TCP

Hi all. We have Snom phones which do have a defined key in order to drop incoming call WITHOUT answering. Pressing that key, a "SIP/2.0 486 Busy Here" message is sent back. We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 or other) which DO NOT have any key to do that (or the key does not work, as is with Siemens C450 IP ): you have to answer and immediatly after hangup the

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Asterisk is behind firewall, I'm able to register with the provider. Calls are coming IN OK, but when I try to call out I got: Got SIP response 603 "Declined" back from 81.15.xx.xx -- #Joseph

Hi everyone, Since three weeks ago, we have been getting A LOT of 603 Declined calls from iCall. I called a few times and their support is either non-responsive (they never call back) or can't fix the issue. I am wondering if everyone else is experiencing the same thing or is it because we recently upgraded from Asterisk 1.6x to Asterisk 1.8x and there is something that is causing this. This