Displaying 20 results from an estimated 1000 matches similar to: "Not getting inbound CallerID name on Asterisk"
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
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.
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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
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
2009 Jul 22
3
CallerPres SIP headers Analog Phone
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?
2014 Jul 26
1
Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
Greetings,
I've noticed a problem that might originate from my Asterisk configuration,
could use a hand in sorting it out. Problem is a 488 response from Asterisk
whenever it gets RTP/SAVPF profile in the SDP.
My current setup has Asterisk Kamailio realtime integration, and Kamailio
uses dispatcher to route calls for Asterisk to handle. Now I have only one
Asterisk, on the same machine as
2010 Nov 03
1
inbound call issue...
Can anyone tell me why my inbound calls keep getting rejected with 401?
Here's the debug information:
<--- SIP read from UDP:147.135.32.221:5060 --->
INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0
Call-ID: 31007e-31 at 147.135.32.221
CSeq: 1 INVITE
From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc
To: "Gregory Malsack"<sip:s at
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
2015 May 28
4
Peer is UNREACHABLE
Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb:
> The phone you gave your wife is really old. Are you sure it supports SIP
> OPTIONS? Can you make a call in or out to it? If you can, it is more
> likely that it just doesn't support that and you can't use a qualify
> statement.
No, I'm not sure.
And no, I can't make any call, right now... At least,
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
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2011 Mar 09
6
SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten => s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
/INVITE sip:0473 at sip.domain.be SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97
From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2
To: <sip:0473 at sip.domain.be>
2017 Aug 17
2
Pass CallerId/Privacy info from A Leg to B Leg
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
2015 Mar 03
6
TLS, SRTP, Asterisk11 and Snom870s
CentOS-6.5 (FreePBX-2.6)
Asterisk-11.14.2 (FreePBX)
snom870-SIP 8.7.3.25.5
I am having a very difficult time attempting to get TLS and SRTP
working with Asterisk and anything else. At the moment I am trying to
get TLS functioning with our Snom870 desk-sets. And I am not having
much luck.
Since this is an extraordinarily (to me) Byzantine environemnt I am
going to ask if any of you have gotten
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
2015 Jun 26
2
Asterisk dialplan best practices syntax
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
2011 Oct 20
1
10.0 CallerID question
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
2012 Aug 17
0
Trouble with call pickup using RPID with Cisco
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:
2016 Apr 23
2
Incoming calls from Andrews & Arnold failing to authenticate
I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk).
VoIPtalk calls are unauthenticated and reach me fine, but Andrews &
Arnold calls are authenticated. The last call I successfully received
was on Tuesday afternoon. Initially, A&A were for some odd reason not
sending calls to my server, but that has been resolved. The problem now
is that the calls fail to
2012 May 08
4
Asterisk 1.8 Transfer CallerID
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