similar to: Weird 'hairpin' call rtp audio problem

Hi Joshua > The "rtp_keepalive" option can be used to have the RTP stack send an > RTP packet out. Try that and see what happens. Once again 'bullseye' that fixed the problem. Thank you! Mit freundlichen Gr?ssen -Beno?t Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the

Hi List I have been pondering over a problem to use an asterisk server behind an SBC unable to successfully handle registrations. Now I observed something strange which I suspect might be a bug on the asterisk side. The SBC originates is register from Port 6011 to Port 5060 on the Asterisk. The Contact Header of the REGISTER contains: Contact: user at SBC-IP:6011 The Asterisk is sending the

Hi List I wonder how SIP via TCP is supposed to work. Not realy Asterisk related, but I hope you experts might be able to help out :-) One of our customers has a SIP device registering via a complex NAT. To benefit from TCP Connection Tracking, he choose TCP instead of UDP. So he expected, that an incoming call would be sent back to him on the already open TCP connection, making it easy to get

Hi Joshua Thank you for your reply. Indeed, Ubuntu only ships with this old version. Upgraded to 16.2. via PPA. Problem persisted. Well, I already mentioned that this is a machine with two physical interfaces with different routes which on the 'external' side handles SIP customer registrations and has an 'internal' IC Trunk to a commercial Voice Switch via private IP Range. I

Hey List I sometimes use our asterisk server to do some debugging or other PBX and SBC. Now we have a case where a PBX is replying an incomming invite with 180 ringing immediately. It looks like the SBC does not accept this. According to my understanding of the RFC 3261 any provisional (aka 1XX) reply should be good enough to make the sender stop re-sending invites and accept this as a reply

Hi Gang To increase security against phished passwords and similar attacks, we consider offering customers to define IP ranges (or GeoIP locations) from which their dynamic registrations are being accepted. I can already look at the source IP in the dial plan, so no issue with validate an INVITE against a source IP. But I would also like to prevent registrations from outside of this

Dear fellow list readers This is the situation: ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP The Patton GW resides on a dynamic IP address, so I cannot really use match=ip in the identify section. The Patton does not send a line parameter. The ISDN Devices behind the patton have different MSN and should be able to send them in the From: Header, so the default endpoint

Hi Richard > That could be possible and would be a bug in chan_sip. Ok, so I switched to PJSIP to see if this behaves differently So ip do a Transfer(PJSIP/${DESTNUMBER}@trunk) And this results in: Failed to parse destination URI '[destnumber scrubber]' for channel PJSIP/trunk-00000011 Do I have to specify the destination number differently when using Transfer with pjsip that I

Dear List I fear I stumbled over a bug in asterisk 13.14.1. My 'phones' are roaming around, sometimes some are connecting from ipv6 enabled networks, another time they are not. If a connection is ipv6 I would prefer to use ipv6 to avoid ipv4-nat problems. I have not specified a transport in the endpoint section, so that the appropriate transport which corresponds to the registration

Hello List Next question where google did not spit out an unsable answer. When redirecting a call with Transfer, I would like to correctly indicate the reason. I did try this: exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)}) exten => XX,n,Dial(SIP/ZZ) exten => XX,n,set(REDIRECTING(reason)=cfb) exten => XX,n,Transfer(SIP/YY) I did try with 'reason'

Hi Richard Thank you > You need to set more redirecting information [1]. > > In sip.conf send_diversion=yes needs to be in effect. You also need > to setup > the from party id information (at least the from number) to indicate > where you > are redirecting from. You should also increment the redirecting > count. > > Richard > > [1] >

Hi Gang Server, two interfaces, routing to two different networks. Two transports defined, each bound to the corresponding ip assigned to the interface. But still, especially when an 183 message is sent, the Contact header does contain the wrong IP Address. Is this a known issue 13.18.3? Or is there a way to make absolutely sure the IP addresses within the Contact header is corresponding to

Hi Joshua I had a shot at your suggestion, bug still no success. I fear the 181 is sent before the macro is called. I want to change the Diversion Header in the 181 message sent back to the caller to put the number it contains in the correct e164 format (stripping the 0 and adding +41 for Switzerland) but just any 'dialplan set' value would do for an example :-) Could you please make

Hi Well it is well some time that my last DUNDI peer has become unreachable. I guess too many issues with spoofed numbers etc. But I am wondering, do people, especially larger entities like telcos, still use DUNDI? I know that in some Hamradio communities, DUNDI is used to interconnect PBXes, but that is with private phone number ranges, not connected to the public. Want some DUNDI peering?

Hi List Implementing screening and routing I have stumbled over this issue: [pbx-router] exten => s,1,NoOp(ROUTER FROM: ${CALLERID(Number)} TO: ${DESTINATION}) same => n,Set(SOURCE=${CHANNEL(name)}) same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) same => n,Set(FROM=${CALLERID(Number)}) same => n,Set(TO=${DESTINATION}) same

Hi Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) In the AGI I do: my $pai = $AGI->get_variable(PAI); This works fine, unless the PAI contains quotes: P-Asserted-Identity: <sip:1000 at 1.2.3.4:5060;user=phone> I get "<sip:1000 at

Hi Gang According to: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk12Configuration_res_pjsip-endpoint_device_state_busy_at And endpoint should return busy if this number is reached. We have PBX Trunks registering to the Asterisk. So we want to limit the number of concurrent calls to a PBX and return busy, if more than the configured number of channels

So long as the tcp socket is open your SBC should send the call back over the same socket. Now it can be that your SBC is seeing the socket as timing out. If you are using Kamailio you can have it send tcp keep alives every so often so that the socket stays up. On Fri, Dec 27, 2019 at 10:41 AM Benoit Panizzon <benoit.panizzon at imp.ch> wrote: > Hi List > > I wonder how SIP via

Hi List One more Problem I stumbled upon. Using Asterisk in a TSP environement. Incomming IC Calls are e164 and have a NPRN (Routing Number) prefixed. Example: +4198055615995555 +41 country prefix 98055 Routing Prefix 615995555 effective phone number Calls routed to Customers need to be put in the 'local' format. 0615995555 This is also the format of the From / To / Invite header