search for: sendrpid

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 attach...

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/asterisk-users/atta...

Hello all If I look at the sip peers table definition as provided with the source of asterisk-1.8.23.0/ (looking at contrib/realtime/mysql/sippeers.sql) for the sendrpid column it's an enum with 2 possible values, yes and no. However, the sip.conf allows 4 values, no, yes, rpid and pai. Is this discrepancy an oversight? Is it possible to set the system default to pai but an individual peer to rpid via a realtime table? I have tried setting the system value t...

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

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

...ip.conf is: [general] language=de port=5060 disallow=all allow=alaw allow=ulaw allow=GSM nat=no canreinvite=no tos=lowdelay context=default [9001] type=friend username=9001 secret=password host=dynamic callerid=Beckenbauer, Franz <9001> context=default mailbox=9001 callgroup=1 pickupgroup=1 sendrpid=yes [9002] type=friend username=9002 secret=password host=dynamic callerid=Walter, Fritz <9002> context=default mailbox=9002 callgroup=1 pickupgroup=1 sendrpid=yes cheers, Sven

...p://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! Thanks, Chris Doug...

Came upon a problem today that I thought I'd see if it's by design, if I'm missing an option somewhere, or if my fix is the way to fix it. We setup a remote location with a server, same as we've done with others, but for some reason when they would transfer an outside call anywhere it would pause for a few seconds and hang up the line. Well, after spending most of the day on

...673581] secret=xyz callerid=Input Interior 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 dire...

...;called party identification" patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the "equivalent" of CALLERID, but the other way...)? BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true. Thanks, __Yehavi:

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 formed by our switch. Thanks in Advance, Nick

hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this?

Hello, when a call comes in and is answered by colleague A, this colleague A sees the CallerID of the external calling number. When colleague A transfers the call to colleague B, attended or unattended, then colleague B sees the number of colleague A on his screen while talking to the external calling number. I expect here that colleague B would see the external calling number on the screen

....,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r) exten => _X.,n,Hangup And here my users.conf: [00493511111111] fullname = luca secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myproxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/00493511111111 [00493512222222] fullname = fax secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no c...

...And here my users.conf: > > [00493511111111] > fullname = luca > secret = MYSECRET > dahdichan = 1 > hassip = yes > hasiax = no > hash323 = no > hasmanager = no > callwaiting = no > context = myproxy > host = dynamic > dtmfmode=rfc2833 > canreinvite=no > sendrpid=pai > type=friend > nat=force_rport,comedia > qualify=yes > qualifyfreq=60 > transport=Auto > avpf=no > force_avp=no > icesupport=no > encryption=no > callgroup= > pickupgroup= > dial=SIP/00493511111111 > > [00493512222222] > fullname = fax > secret =...

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

...39;00493513333333' rejected because extension not found. users.conf on Ubuntu-PBX: [00493511111111] fullname = 00493511111111 secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/00493511111111 [00493512222222] fullname = 00493512222222 secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasman...

Hi All I have one issue regarding override callerid when i have anonymous call. I have added PAI in sip header and also set sendrpid = yes in sip.conf but the callerid is not overriding while i am sending call to three digit calling like 911. please give some idea and help for this issue! I am using asterisk 1.4 branch. thanks in advance!! Thanks, Max Alex Voip Developer -------------- next part -------------- An HTML attachmen...

When you set sendrpid=yes in sip.conf, a very nice feature is activated. When dialing an extension, the callerid of the dialed extension is returned back on the display of the calling phone. So if you call extension 100, you can see you are calling Ann (for example). I want to selectively disable the transmission of th...

...ster => 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= pickupgroup...