search for: trustrpid

Hi everyone, I am trying to figure out the behavior of trustrpid Basically its not behaving the way I expected it to or maybe I am missing a configuration option or something else. When a call from a phone is sent to the * box it has the following sip headers: From: "From Phone" <sip:1001 at 10.0.0.29>;tag=4bf4bb4e11e92476. Remote-Party-ID:...

Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-00000200 prevented Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asteri...

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/atta...

Hi Are there any security issues I need to be aware of if I set trustrpid to yes in my sip.conf? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

...Orebro (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=yes context=inputinterior.se directmedia=yes dtmfmode=rfc2833 faxdetect=no h...

...d both signed and self-signed cert to no avail. Here is my Configuration: Sip.conf tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/box1.pem tlscapath=/etc/asterisk/keys tlscipher=ALL tlsclientmethod=tlsv1 sip.conf ext. [5006] type=peer context=sipext call-limit=3 trustrpid=no callerid="Rec" <5006> disallow=all allow=ulaw allow=alaw username=5006 secret=9fcbb025200881850526bc57d59885c3 dtmfmode=rfc2833 host=dynamic mailbox=5006 nat=yes canreinvite=no transport=tls == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_...

...y upgraded from Asterisk 1.8 to Asterisk 13. Now users are starting to complaint about packets loss, conversations are choppy! I don't even know where to start looking! Choppy conversations happened within users. I am using sip.conf [1091] type=friend context=sip-phone call-limit=2 trustrpid=no callerid="dev1" <1091> disallow=all allow=ulaw allow=alaw username=1091 secret=XXXXX dtmfmode=rfc2833 host=dynamic mailbox=10091 at default nat=force_rport,comedia canreinvite=no extensions.conf exten => 1091,hint,SIP/${EXTEN} exten => 1091,1,Dial(SIP/${EXT...

Hi, I've two yocto questions about the syntax of dialplan: 1. What's the "official" notation of each line: "=>" or "=" ? In the wiki of Asterisk, I see very often "=>", however, what's the reason for both syntaxes authorized ? Historical ? 2. To write info in logs/console, you have two commands: NoOp and Verbose. Verbose seems to be

hello, All AreskiCC users: I faced some problems in using AreskiCC. one is when I reload the asterisk server, the system display some errors such as execution 30 .. second one is there is no data display for admin added before. Does anyone know how to solve the problems, Please tell me! thanks in advance!

Hello Everyone, Our switch is sending P-Asserted info to asterisk however the information is getting removed when asterisk forks the call. Is it possible to have asterisk retain the P-Asserted on the leg. This is quite important for valid functionality of our network. Tried setting `sendrpid = yes` and still same problem. We really don't want to have to `SipAddHeader` as it is already being

...ip.conf: register => 50607795:test at 10.10.33.228/50607795 register => 50607796:test2 at 10.10.33.228/50607796 [50607795] accountcode=mobiltest defaultuser=50607795 type=peer host=10.10.33.228 canreinvite=no insecure=port,invite context=from-inside secret=test fromuser=50607795 trustrpid=yes sendrpid=yes [50607796] accountcode=mobiltest defaultuser=50607796 type=peer host=10.10.33.228 canreinvite=no insecure=port,invite context=from-inside secret=test2 fromuser=50607796 trustrpid=yes sendrpid=yes On the "server", these are configured: [50607795] callgroup...

...BLACKLIST()}?black,1) exten => _X.,n,Ringing exten => _X.,n,Progress() exten => _X.,n,Wait(5) exten => _X.,n,Dial(SIP/123&SIP/456,30,oxX) ... exten => fax,1,NoOp(**** FAX DETECTED ****) exten => fax,n,Goto(fax-rx,receive,1) in the sip.conf i specified [general] sendrpid=rpid trustrpid=yes language=de videosupport=yes callevents=yes caninvite=yes qualify=yes nat=force_rport,comedia faxdetect=yes t38pt_udptl=yes ... [abcde] type=peer insecure=invite defaultuser=12345678912 fromuser=12345678912 fromdomain=abcde.ab secret=guess-what host=abcde.ab qualify=yes context=from-sip dtmfm...

...equest is processed within a particular window of time. For this to occur, the following must take place: 1. The setting 'trustrpid' must be set to True 2. An UPDATE request must be received after a call has been terminated and the associated channel object has been destr...

Hi List, Another dumb conversion question (I hope). I installed 10.0 and copied my 1.4 configuration files over. With a few tweaks everything works great except for 1 feature that I specifically went to 10.0 for. When I do an attended transfer, I still get the receptionists caller ID on the transferred phone instead of the incoming callerID. My assumption is that there is some

...e phones. http://pastebin.com/m45e0adbd Here is the section from Sip.conf describing the Cisco 3825 connection. We have tried "type" as both friend and peer as it is now with no change. [cisco_3825] context=default type=peer host=10.0.0.10 disallow=all allow=g729 allow=ulaw allow=alaw trustrpid=yes sendrpid=no All phones are not receiving the CallerID name, here is a sample from sip.conf of a phone config. [8670] secret=8670 context=ict_sip type=friend host=dynamic call-limit=5 agentlogin=yes mailbox=8670 at ictvm progressinband=no sendrpid=yes Any help is greatly appreciated! Thank...

...for each server: PBX01 [pbx01topbx02] type=friend context=incomingDefault host=10.10.10.2 qualify=600 disallow=all allow=ulaw canreinvite=no insecure=invite accountcode=pbx02 ;TESTS; ;auth=pbx01topbx02 at asterisk ;fromuser=pbx01topbx02 ;username=pbx01topbx02 ;secret=j48dj7rjd9023jd sendrpid=yes trustrpid=yes PBX02 [pbx01topbx02] type=friend host=10.10.10.3 qualify=600 context=dialOutPatternsAll disallow=all allow=ulaw canreinvite=no insecure=invite accountcode=pbx01 ;TESTS; ;auth=pbx01topbx02 at asterisk ;username=pbx01topbx02 ;fromuser=pbx01topbx02 ;secret=j48dj7rjd9023jd sendrpid=yes trustrpid...

...sip.conf snippet from a popular provider's web site: [xxx-inbound] type=friend dtmfmode=auto host=xxx.yyy.zzz context=inbound username=xxx secret=yyy allow=all insecure=port,invite canreinvite=no [xxx-outbound] type=friend dtmfmode=auto host=xxx.yyy.zzz username=xxx fromuser=xxx secret=xxx trustrpid=yes sendrpid=yes allow=all canreinvite=no Pretty ugly and difficult to read. With a little whitespace and alphabetizing we get: [xxx-inbound] allow = all canreinvite = no context = inbound d...

...p address of the peer. What is the solution for this problem? How can asterisk update the peer? The Asterisk is local behind a NAT with a firewall, following settings are used: externhost with DynDNS stun with stun.t-online.de <http://stun.t-online.de/> nat=yes srvlookup=yes allowguest=no trustrpid=no insecure=invite qualify=yes Thank you! Daniel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150330/9cb4cb90/attachment.html>

Hello, using Asterisk 1.8.12.2 case : I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the