search for: rpid

I have a Cisco 1760V with FXO ports hooked to POTS lines talking SIP to asterisk 1.8.15.0. imagining in extensions.conf: exten => 1,1,Dial(SIP/121) exten => 2,1,Dial(SIP/121&SIP/122) When a caller dials extension 2 /and/ I have trustrpid=yes generaterpid=yes sendrpid=yes in sip.conf and I use the pickup exten, the caller is disconnected. see: http://jeremy.kister.net/tmp/ast/group-with-rpid if i set the rpid generate/send = no for the cisco peer, the user is connected. see: http://jeremy.kister.net/tmp/ast/group-without-rpid...

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

...manage to force asterisk to put Remote-party-ID attribute on the SIP outgoing call? I.e. When A calls B, I want that A gets a name of B displayed on his phone. Note that name of A gets displayed on the B's phone fine, but this is not what I want. This works with Cisco Call manager fine - the RPID is sent as a part of the response to the SIP INVITE this way: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport From: "Ondrej Valousek" <sip:7775 at 192.168.60.20> <sip:7775 at 192.168.60.20> ;tag=as4786d518 To: <sip:1098 at 192.168.62....

We have an inbound PRI connected to our Cisco 3825 router which is then passing the calls to Asterisk as SIP calls. We're getting the CallerID number but not the CallerID name. We are seeing the name in the RPID field with a SIP trace on the Asterisk box but don't understand why it's not registering as the CallerID name. Here is a link to pastebin with the Sip trace. In it you can see the RPID is seen from the Asterisk box but it is not used/sent to the phones. http://pastebin.com/m45e0adbd H...

Hello, I am looking for a softphone which supports RPID (displaying the called party name) and BLF features. I couldn't find one so far... Any idea whether such a softphone exists? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/aster...

Hello, I am running Asterisk 1.4 with the RPID patch (8824) in order to display the name of the *called* person. It works on Cisco and Polycom phones, but not on SNOM. The local SNOM dealer claims that he saw it working, but he cannot give me more details. Has anyone managed to make it working? Thanks, __Ye...

...llow=ulaw insecure=invite,port context=from-trunk secret=password nat=yes username=<tgid> with a registration at: <tgid>@<sip domain>:password:<tgid>@siptrunk/<tgid> Using the above configuration, the trixbox successfully registers to the BroadWorks. I also enabled RPID in the Asterisk. Each of the four SIP phones is registered with the Trixbox and are set to use the siptrunk for outbound calls. So, outbound calls (SIP/Asterisk -> PSTN/Broadworks) work properly. The RPID is set to the TGID and the From is set to the CLID from the SIP phone user. Broadworks...

maybe usefull for displaying CALLED party name when dialing.... I'm remember, that this feature was planned to add to asterisk, any progress? PJ New and Changed Information Release 8.0(0) includes the following new and enhanced features: ?Remote-Party ID support has been added for incoming INVITE and UPDATE requests and 18x and 200 responses.

...box : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : "" <> Expire : 113 Insecure : no Nat : RFC3581 ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Trust RPID : No Send RPID : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 10.4.5.1 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Sock fd : 24 Transport : TCP Def. Username: 971 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order...

...24 Insecure : no Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : Yes TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 1.1.1.1:62850 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP De...

...xpire : -1 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 201.217.31.10 Addr->IP : 201.217.31.10:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 59521...

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? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090722/0564e30d/attachment.htm

...uot; <sip:8005551234 at xxx.xxx.xxx.xxx>;party=calling;privacy=off;screen=no For the AMI Originate, I have been passing variables in an attempt to modify the CALLERID(name-pres). My understanding is that a variable of "CALLERID(name-pres)=allowed_passed_screen" should result in the RPID screen setting being yes. I have tried many different values for this variable, but the RPID line is always "screen=no". What am I missing to force the screen=yes to be passed as part of the Remote-Party-ID? Dan -------------- next part -------------- An HTML attachment was scrubbed.....

...t;> T.38 support : No >> T.38 EC mode : Unknown >> T.38 MaxDtgrm: 4294967295 >> DirectMedia : No >> PromiscRedir : No >> User=Phone : No >> Video Support: Yes >> Text Support : No >> Ign SDP ver : No >> Trust RPID : No >> Send RPID : Yes >> TrustIDOutbnd: Legacy >> Subscriptions: Yes >> Overlap dial : No >> DTMFmode : rfc2833 >> Timer T1 : 500 >> Timer B : 32000 >> ToHost : >> Addr->IP : 1.1.1.1:62...

...presence Accept: application/pidf+xml Content-Length: 511 Content-Type: application/pidf+xml Expires: 3600 User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) <?xml version="1.0" encoding="UTF-8"?> <presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:pidfonline="http://www.linphone.org/xsds/pidfonline.xsd" entity="sip:john at xxx.xxx.com" xmlns="urn:ietf:params:xml:ns:pidf"> <tuple id="bhhmlg"> <status> <basic>open</basic> &l...

...secure : no Force rport : Yes Symmetric RTP: Yes ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID : No TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 192.168.11.160:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Use...

...re : 3326 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : Yes Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : x.x.x.x:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: PACK501 SIP Options...

...x : 245 at device VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 50 Dynamic : Yes Callerid : "device" <245> MaxCallBR : 384 kbps Expire : 67 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 192.168.0.239 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 245 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing: No Status : OK (124 ms) Userag...

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, I configured both asterisk and grandstream 2000 accourding to howtos on the web.. And everything seems working fin. But if i reload asterisk grandstream stops working with BLF. I need to restart the phone to enable BLF again. Any clues??