similar to: Re: Server freeze with meetme and sip GSM users

Hi Brent > Anyone ever seen MeetMe cause * to crash? Specifically, it happens > consistantly if someone begins to enter a conference and then decides to > hangup while Allison is introducing them - like playing back > "conf-onlyperson". This has been seen with the MeetMe participant > connecting via IAX and SIP (not saying it doesn't happen with Zap, just > that I

Dollars to donuts it is related to these two posts, but no one seems to know where or why it happens - this issue doesn't seem to be related to one specific piece of hardware: Post 1) ********************************************************* Anyone ever seen MeetMe cause * to crash? Specifically, it happens consistantly if someone begins to enter a conference and then decides to hangup while

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 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 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

Hello Le 04/12/2023 à 10:56, Benoit Panizzon (by way of Benoit Panizzon <benoit.panizzon at imp.ch>) a écrit : > Hi List > > We have some CPE which run an embedded asterisk 13 with chan_sip. > > Unfortunately, when a registration is rejected, those stop trying. > > I am familiar with pjsip which allows to configure: > > auth_rejection_permanent=no > > How

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the

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 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 All I'm tracing a very strange problem which I could reproduce with different versions up to 1.2.5 (sorry, didn't update to a newer one yet). Scenario 1: Problem does not occure. ============================= Sip Phone registered directly to the Asterisk. exten => i,1,Zapateller() exten => i,n,Playback(invalid,noanswer) exten => i,n,Hangup() Works like expected. I dial an

Hi all I have a problem that starts driving me crazy... Win2k3 ADS, added some attributes like loginshell, gid, uid etc. Unix clients use NSS_LDAP to get 'passwd' data and kerberos to authenticate users. Authentication does not happen via LDAP. winbindd is used to autocreate sid => uid/gid mappings. This worked very fine with samba 3.0.14a. Upgraded to samba 3.0.23 Now the owner

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

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

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

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] >