similar to: Getvar of CHANNEL not working for a couple of items

Displaying 20 results from an estimated 100 matches similar to: "Getvar of CHANNEL not working for a couple of items"

2023 Jul 05
1
Getvar of CHANNEL not working for a couple of items
Channel A: "1688509741.112" , name: "PJSIP/111-00000064" , is originator: Y , call-Id: "u.l6kcou25cax60 at mydomain.com <mailto:u.l6kcou25cax60 at mydomain.com> " , local_uri: "<sip:222 at mydomain.com <mailto:sip%3A222 at mydomain.com> ;user=phone>" , local_tag: "1734d973-c4da-4ae8-a37d-5f7065f1fe54" , local_addr:
2023 Jul 05
1
Getvar of CHANNEL not working for a couple of items
On Tue, Jul 4, 2023 at 7:52 PM TTT <lists at telium.io> wrote: > Building on my last message, I am trying to get CHANNEL data using getvar > (through the AMI). And although I'm getting responses, some values > returned seem illogical. For example, phone 111 calls phone 222 via the > PBX. Here's the data I get back > > > > > > Channel A:
2023 Jul 04
1
Getvar of CHANNEL not working for a couple of items
Building on my last message, I am trying to get CHANNEL data using getvar (through the AMI). And although I'm getting responses, some values returned seem illogical. For example, phone 111 calls phone 222 via the PBX. Here's the data I get back Channel A: "1688509741.112" , name: "PJSIP/111-00000064" , is originator: Y , call-Id: "u.l6kcou25cax60 at
2023 Jul 04
1
Getvar of CHANNEL not working for a couple of items
The following AMI command works well for all of the information I want: action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-3-In-00000028 Variable: CHANNEL(pjsip,XXXX) Where XXXX can be one of the many available items. However, when I create a call from A to B, all of the items return properly except: local_addr and remote_addr. More specifically, they return correctly for the A leg (that
2023 Jun 26
1
Get channel variables via ARI/AMI
On Mon, Jun 26, 2023 at 4:35 PM TTT <lists at telium.io> wrote: > I think that’s getting me close. I’m trying to get (or recreate) the FROM > and TO lines of the header, from a system running PJSIP. I think if I use > CHANNEL to get local_uri and local_tag I can recreate a FROM line like: > > *FROM=<URI>;tag=TAG* > > > > And if I use CHANNEL to get
2023 Jun 26
2
Get channel variables via ARI/AMI
I think that’s getting me close. I’m trying to get (or recreate) the FROM and TO lines of the header, from a system running PJSIP. I think if I use CHANNEL to get local_uri and local_tag I can recreate a FROM line like: FROM=<URI>;tag=TAG And if I use CHANNEL to get remote_uri and remote_tag I can recreate a FROM line like: TO=<URI>;tag=TAG Would it be correct to assume
2011 Sep 05
1
Variables error in 1.8.6.0.
Hello, I have a problem with some variables in 1.8.6.0. I set on extension the following lines: exten => h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio, local_lostpackets)}) ; lost packets by local end ** exten => h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio, remote_lostpackets)}) ; lost packets by remote end exten => h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos,
2015 Sep 23
2
problems with PJSIP install on UBUNTU 14.04
Ok that did it after I did the steps to completely remove everything and do a new install. Thanks! > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Joshua Colp > Sent: Wednesday, September 23, 2015 10:12 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject:
2015 Sep 23
2
problems with PJSIP install on UBUNTU 14.04
Ok so now I'm getting this when doing a make in asterisk... travis at pcimphone1:~/downloads/asterisk-13.5.0$ make [LD] chan_pjsip.o pjsip/dialplan_functions.o -> chan_pjsip.so /usr/bin/ld: /usr/local/lib/libpjsip-ua-x86_64-unknown-linux-gnu.a(sip_inv.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC
2023 Jun 26
1
Get channel variables via ARI/AMI
On Mon, Jun 26, 2023 at 4:04 PM TTT <lists at telium.io> wrote: > It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the > entire SIP header for a channel. I also read (on stackoverflow) that the > PJSIP_HEADER function will only return the headers from the INVITE of the > *inbound* channel. > > > > If that’s correct, how would I get the headers from
2004 Jul 21
2
fonction Getvar
Hia .... i try to use the fonction Getvar of asterisk to get a variable myDNIS that i have define. i use it as follow Action: Getvar Channel: SIP... Variable: myDNIS but asterisk don't know it .i have the response as follow Response: Error Message: Invalid/unknown command does everybody meet this problem . i try all possible combination and nothing help please ..!! :-( thanks in advance
2005 Jul 12
0
Manger-command Getvar?
Hi, I'm trying to use the manager cmd Getvar. Unfortunately I always get (null) as variable content. I'm using asterisk 1.0.7 When calling a non existant channel, I get an appropriate result. This is what I tried and got: Action: Getvar Channel: SIP/01234567-5242 Variable: CALLERID Response: Success CALLERID: (null) Any hints? Roger.
2023 Jun 21
1
Asterisk not replacing private FROM ip with public IP in INVITE
Something perhaps noteworth, since this is a multihomed system I bound the transport to 172.31.253.4:5060 I don't *think* that would cause Asterisk to use that IP in the FROM...at least it shouldn't. -----Original Message----- From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of TTT Sent: Wednesday, June 21, 2023 2:58 PM To: 'Asterisk Users Mailing
2023 Jun 21
2
Asterisk not replacing private FROM ip with public IP in INVITE
I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: From: "MYNAME" <sip:16667778888 at 172.31.253.4>;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4 The IP address above is an internal/non-routable IP, so Twilio is rejecting it. For some
2008 Sep 12
2
SCCP port numbers used for audio stram?
I have a 7921 wireless phone working with Asterisk, and I want to tighten the wide open port range of my IPTABLES now. I tried allowing only SCCP port (2000) in/out and found that my audio was gone. A quick look at my iptables message shows source port 15886 and dest port 25968 used: FORWARD - Drop: IN=eth1 OUT=eth2 SRC=172.31.253.4 DST=172.31.254.102 LEN=200 TOS=0x18 PREC=0xA0 TTL=63 ID=0
2023 Jun 21
1
Asterisk not replacing private FROM ip with public IP in INVITE
I tried that (only needed to add rewrite_contact=yes) but it didn't help. BTW, the CONTACT: line holds the correct ip! Only the FROM: line holds the wrong (private) IP. I'm still learning SIP...but I assume the FROM should also hold the rewritten public IP. Just don't know how to force Asterisk to do that. -----Original Message----- From: Eric Wieling [mailto:ewieling at
2008 Nov 28
1
Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call
I'm trying to get my Windows Mobile 6 phone working as an asterisk client. Overall things are working well. However, I regularly get the following message: [Nov 27 21:57:28] WARNING[4507]: chan_sip.c:12892 handle_response: Remote host can't match request NOTIFY to call '67aa8e223479055656161bf17ebb77d5 at 172.31.253.4'. Giving up.
2023 Jun 21
1
Asterisk not replacing private FROM ip with public IP in INVITE
type=endpoint rewrite_contact=yes force_rport=yes rtp_symmetric=yes On 6/21/23 14:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: > > From: "MYNAME" <sip:16667778888 at
2023 Jun 21
1
Asterisk not replacing private FROM ip with public IP in INVITE
You need to put your external IP in the transport configuration: external_media_address=X.X.X.X external_signaling_address=X.X.X.X external_signaling_port=5060 On 21/06/23 12:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction:
2007 Jul 12
0
No subject
event - which it rejects. The voicemail notifications ARE working on the device. Any way to get rid of this message (while keeping the MWI on the phone)? ------=_NextPart_000_0072_01C950DB.BAD4D4E0 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <html xmlns:v=3D"urn:schemas-microsoft-com:vml" =