similar to: No ID from the calling party in SIP Header

Displaying 20 results from an estimated 1000 matches similar to: "No ID from the calling party in SIP Header"

2007 May 03
2
Called party identification - where to take called name?
Hello, I am trying to apply the "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.
2014 Sep 05
2
Asterisk with PJSIP
Hi All, I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on CentOS7. -- https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject The installation is OK. But the connected SIP cilents (both Linphone on Windows7) cannot communicate. I hope your comment such as the testing for resolving the problem. My status is the following(1 and 2). Why 'Everyone
2011 Jan 10
3
sendrpid does not work!
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:
2006 May 26
1
Not able to make any calls
Hi All, I have registered "abhijit" for SIP in asterisk Server. I am able to register my softphone (SJPhone) to the server using the name "abhijit". But whenever I try to make any calls I am gettinh the following error message:- *CLI> -- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper:
2013 Feb 15
6
Cisco 7942 Connected line 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:
2009 Jul 26
3
Not getting inbound CallerID name on Asterisk
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
2010 Feb 20
1
Fax, T38 and NAT
Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38
2007 Aug 24
0
[Fwd: Re: issues with caller ID , remote-party-id
Hello ppl, Sorry to re-post it, but kinda these issues are getting on my nerves. I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on 1.4.4. The problem : 1. I receive call from caller 'AAA' on my number, 'BBB' which is on my Asterisk box. 2. I have to redirect the call to some other number, say, my cell num - 'CCC'. 3. My PSTN provider wants the
2014 Feb 16
1
Retaining P-Asserted Info
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
2015 May 28
3
Peer is UNREACHABLE
Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>
2012 Aug 13
1
Websockets on Asterisk 11 and SipML5
Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets I added transport=ws to my sip.conf file: [3002] username=3002 secret=XXXXXXXXX host=dynamic type=friend context=test disallow=all allow=g729 ;allow=all ; Allow codecs in order of preference allow=ilbc
2010 Apr 01
3
RPID on called party
Hello, Did anyone 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
2006 Jul 25
3
problem with *.foobar.com urls
I am using Apache2.2 + mod_proxy_balancer + mongrel setup and my mongrel.conf file(for apache is like this) 2 3 <Proxy balancer://mongrel_cluster> 4 BalancerMember http://127.0.0.1:9000 5 BalancerMember http://127.0.0.1:9001 6 BalancerMember http://127.0.0.1:9002 7 </Proxy> 8 9 <VirtualHost *> 10 ServerAdmin rags at
2008 Nov 28
1
Anonymous callerid
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
2013 Oct 03
1
Disable the Connected Line info
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 this information back to the caller. How can I do it? I tried setting
2010 May 06
2
problem with 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: "Cloutier"
2014 Jan 21
3
Asterisk Fax detection *11.7
Hello everybody I'm trying to enable the Digium res_fax app at my *11.7 Server. a fax show stats comes up with FAX Statistics: --------------- Current Sessions : 0 Reserved Sessions : 0 Transmit Attempts : 0 Receive Attempts : 1 Completed FAXes : 1 Failed FAXes : 1 Digium G.711 Licensed Channels : 1 Max Concurrent : 0 Success : 0 Switched to
2010 Jul 01
3
Remote Party ID issue
Hi, i have the same problem. Trying to use the dialplan function CONNECTEDLINE() this way Set(CONNECTEDLINE(name)=${SIPPEER(${EXTEN},callerid_name)}) Set(CONNECTEDLINE(num)=${EXTEN}) ends with [Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function CONNECTEDLINE not registered Same happens trying function CALLEDID. I am using Asterisk 1.6.1.20. What do i
2014 Apr 16
1
Connecting 2 asterisks, one with PJSIP and other SIP returning 401
It's my first post here, so I'll cut to the chase I have 2 Asterisk servers and want to connect them using sip on one and pjsip on the other one. One is running at home and another at a VPS. The first one will be the client (with dynamic ip) and the 2nd the server. The client uses sip and the server pjsip. This is the client's sip.conf [general] context = default allowguest = no
2013 Feb 04
1
CallerID external call after Attended Transfer
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