Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

23.04.2019 0:27, Joshua C. Colp wrote: On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote: Tried already. "line" is good, but not perfect. Every time I restart asterisk, it will generate new random string for ";line=". So, every time I restart asterisk, registrar (Server1) will save one more

Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote: > Tried already. > > "line" is good, but not perfect. > > Every time I restart asterisk, it will generate new random string for > ";line=". > > So, every time I restart asterisk, registrar (Server1) will save one > more contact in it's

Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121.

Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: > Hi, > > Got problems with incoming SIP calls. > > Scenario: > > Server1: 3cx or any other server > > Server2: Asterisk 16.2.1 . PJPROJECT 2.8 > > Server2 registers on Server1 with SIP ID 1121. > > Registration is OK. > > Server2 outgoing

[asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,...

Re: [asterisk-users] incoming call label

Hello Thelma, Friday, February 16, 2018, 2:16:02 AM, you wrote: > Contact: "sip:pstn-" > And it found in sip.conf only: > Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 > Is perhaps the name effected by the special character "-" (dash) that is > why it only matches "pstn" and

Re: [asterisk-users] incoming call label

Le 16/02/2018 à 05:30, the...@sys-concept.com a écrit : On 02/15/2018 04:49 PM, Joshua Colp wrote: On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: Thanks again for the hint. Here is the output from asterisk. The call is coming on Audocodes gateway from: pstn- But

Re: [asterisk-users] incoming call label

On 02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn- >> >> But asterisk display: >> Found peer

Re: [asterisk-users] incoming call label

Thelma On 02/15/2018 07:16 PM, the...@sys-concept.com wrote: > > On 02/15/2018 04:49 PM, Joshua Colp wrote: >> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: >> >> >> >>> >>> Thanks again for the hint. >>> Here is the output from asterisk. >>> >>> The call is coming on

Re: [asterisk-users] incoming call label

On 02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn- >> >> But asterisk display: >> Found peer

Re: [asterisk-users] incoming call label

On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > Thanks again for the hint. > Here is the output from asterisk. > > The call is coming on Audocodes gateway from: pstn- > > But asterisk display: > Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 > > Why not

Re: [asterisk-users] incoming call label

On 02/15/2018 04:08 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote: >> On 02/15/2018 03:44 PM, Joshua Colp wrote: >>> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote: I'm using Audio-codes MP-114 unit and it has two public lines PSTN

Re: [asterisk-users] incoming call label

On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote: > On 02/15/2018 03:44 PM, Joshua Colp wrote: > > On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote: > >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports > >> > >> IN audocodes setting I have: > >>

Re: [asterisk-users] incoming call label

On 02/15/2018 03:44 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote: >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >> >> IN audocodes setting I have: >> "EndPoint Phone Number" >> >> Channel: 3phone number: pstn- >>

Re: [asterisk-users] incoming call label

On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote: > I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports > > IN audocodes setting I have: > "EndPoint Phone Number" > > Channel: 3phone number: pstn- > Channel: 4phone number: pstn-9998 > > When I am

[asterisk-users] incoming call label

I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports IN audocodes setting I have: "EndPoint Phone Number" Channel: 3phone number: pstn- Channel: 4phone number: pstn-9998 When I am calling " pstn-" the port number "Channel:3" lights up but asterisk is showing

Re: [asterisk-users] Incoming Call by DID

On Wednesday 26 Oct 2016, KyD wrote: > Hi, > > My sip provider gave me 2 numbers for the incoming call via pstn. > > nro1 = 12341234 > nro2 = 45674567 > > I have a dialplan for each. > if i put this on my dialplan: > > exten => s,1,Dial(SIP/1001) > exten => Hangup() > > Works! > > But if i

Re: [asterisk-users] Incoming Call by DID

It seems like your SIP provider is not sending and DID information, or that the information is not being sent in the same format you are using in your dialplan. You can check this by looking at the SIP debug information for the inbound calls and/or by checking with your SIP provider (that they

[asterisk-users] Incoming Call by DID

Hi, My sip provider gave me 2 numbers for the incoming call via pstn. nro1 = 12341234 nro2 = 45674567 I have a dialplan for each. if i put this on my dialplan: exten => s,1,Dial(SIP/1001) exten => Hangup() Works! But if i put one of them: exten => 12341234,1,Dial(SIP/1001) exten =>

Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

Hello Phil, On Saturday, April 23, 2016, 11:11:29 PM, you wrote: > Actually, this is now sorted. It turns out the latest recommended > configs on the A wiki had peer vs. user confusion. On correcting > this, all was well. I'm glad you found it. It look me a while to track down that problem when

Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

On Sat, 23 Apr 2016 22:45:32 +0100 Julian Beach wrote: > Hello Phil, > > I have a couple of lines with A, and I have not been having any > problems recently. When I have had similar problems in the past, it > has been an issue with the SIP config. I originally had a number

Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

Hello Phil, On Saturday, April 23, 2016, 12:19:15 PM, you wrote: > I have checked that the username and password in my config agree both > ends, and have even tried changing them. > The bulk of my calls come in on A, so I am obviously trying to find > out what has gone wrong. No-one else is

[asterisk-users] 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 were for some odd reason not sending calls to my

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

On Sun, 20 Mar 2016, Trey Hilyard wrote: On Mar 18, 2016 8:27 PM, "Steve Edwards" wrote: >> >> On Fri, 18 Mar 2016, Trey Hilyard wrote: >> >>> I thought this would be as easy as >>> exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10}) > > > How about

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

On Mar 18, 2016 8:27 PM, "Steve Edwards" wrote: >> >> On Fri, 18 Mar 2016, Trey Hilyard wrote: >> >>> I thought this would be as easy as >>> exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10}) > > > How about something like: > > [parse-lrn] >

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

On Fri, 18 Mar 2016, Trey Hilyard wrote: I thought this would be as easy as exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, - Steve Edwards

[asterisk-users] Incoming INVITE with Portability Info and LRN

I am trying to set up my Asterisk server so that it will recognize an incoming call to the Asterisk's own Location Routing Number (LRN), validating the "rn" in the INVITE and then using the Called Number from the INVITE as the extension in the dialplan. The INVITE R-URI looks like: INVITE

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI wrote: > Le 18/03/2016 16:20, Trey Hilyard a écrit : > > I am trying to set up my Asterisk server so that it will recognize an > > incoming call to the Asterisk's own Location Routing Number (LRN), > > validating the "rn" in

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

Le 18/03/2016 16:20, Trey Hilyard a écrit : I am trying to set up my Asterisk server so that it will recognize an incoming call to the Asterisk's own Location Routing Number (LRN), validating the "rn" in the INVITE and then using the Called Number from the INVITE as the extension in the

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

I thought this would be as easy as exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10}) But it appears that the pattern match doesn't work once I get to the "r" in "rn". I am assuming that the pattern match doesn't like dealing with characters without taking the entire URI. I

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

On Fri, 18 Mar 2016, Steve Edwards wrote: Have you tried the '_!.' pattern? The '_x.' pattern works fine. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

On Fri, 18 Mar 2016, Trey Hilyard wrote: I thought this would be as easy as exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10}) How about something like: [parse-lrn] exten = _x.,1, verbose(1,[${EXTEN}@${CONTEXT}]) same = n,

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Larry Moore omninet.net.au> writes: > > sip.conf > > [general] > faxdetect=t38 > > [sipcall.ch] > directmedia=no > > In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a > T.38 re-invite this will trigger the switch to the Fax extension. > > If this proves successful you

[asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around

Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

El 20/01/16 a las 16:25, Alex Villací­s Lasso escribió: I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4

Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

El 20/01/16 a las 18:33, Alex Villací­s Lasso escribió: El 20/01/16 a las 16:25, Alex Villací­s Lasso escribió: Partial fix: Google Chrome accepts the call if videosupport is set to "no". This is the SDP of the successful INVITE that Chrome accepts: INVITE

Re: [asterisk-users] Incoming calls get 488 error

On 8/21/15 6:45 PM, Technical Support wrote: I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset. I captured the SIP traffic and see that my M65 is replying with an 488 not acceptable here. From what I read this is usually codec related but

[asterisk-users] Incoming calls get 488 error

I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset. I captured the SIP traffic and see that my M65 is replying with an 488 not acceptable here. From what I read this is usually codec related but both asterisk and the M65 are set for ulaw as

Re: [asterisk-users] Incoming calls get 488 error

- From: Technical Support [supp...@telium.ca] Received: sexta-feira, 21 ago 2015, 19:46 To: asterisk-users@lists.digium.com [asterisk-users@lists.digium.com] Subject: [asterisk-users] Incoming calls get 488 error I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring

[asterisk-users] Incoming calls to a GSM gateway SIP/2.0 401 Unauthorized response when dial 7777 to Asterisk

Hello: I'm newbie in asterisk, please help me. My context is as follows: 192.168.4.2 -- Asterisk 11.13.1 complied from source 192.168.4.4 -- Yeastar NeoGate TG100 GSM gateway When I call from a GSM cell phone, my TG100 GSM gateway answers and dials extension (configured as a hotline on

Re: [asterisk-users] incoming calls fall into echo test mode

On Saturday 19 Jul 2014, Norman Molhant wrote: I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with

[asterisk-users] incoming calls fall into echo test mode

Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has this problem: his outgoing calls are ok, but when someone dials his phone number (be it from our network or

Re: [asterisk-users] incoming calls fall into echo test mode

is being misrouted in the dialplan From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Norman Molhant ad...@csur.ca Sent: Saturday, July 19, 2014 10:43 AM To: Asterisk Users List Subject: [asterisk-users

Re: [asterisk-users] incoming calls fall into echo test mode

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant Sent: Saturday, July 19, 2014 10:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming calls fall into echo test mode Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each

Re: [asterisk-users] incoming calls fall into echo test mode

To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming calls fall into echo test mode Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote: . . . [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret=gueswhat host=voipdomain.com qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=123456789 add

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Thanks Am 03.02.2014 11:57, schrieb Larry Moore: On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote: . . . [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote: Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Hmm, the fax will be received as an audio call rather than T.38, setting t38pt_udptl=no has turned off T.38. Do you know if your upstream provider supports T.38?

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]:

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55]

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

On 3/02/2014 8:42 PM, Jakob-Matthias Böttger wrote: Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working

[asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Hi, im using a Asterisk Server which is not behind NAT. First i had problems with the fax detection. But this is now solved after adding a wait(2) at the correct place. But i'm still unable to receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short after the Fax session

[asterisk-users] incoming DAHDI Channel explained

Hi, I use a Sangoma A104d-Card (with 4 x germany E1). I process some calls via an AGI-Script. When parsing the AGI-Variables I can see one that look like that: [agi_channel] = DAHDI/i3/211123456-89c What hat do the values mean in detail, please? DAHDI : this is clear i3 : does it mean,

Re: [asterisk-users] incoming DAHDI Channel explained

Sangoma's tech support is probably the better source of information. DAHDI: obviously DAHDI channel i: incoming call 3: span 3 (not the port) 211123456: CLID, probably subject to filtering (see national/international prefix settings) 89c: internal counter (i.e. 2204 calls so far) jg --

Re: [asterisk-users] incoming DAHDI Channel explained

Sangoma's tech support is probably the better source of information. DAHDI: obviously DAHDI channel i: incoming call The 'i' is for ISDN not incoming call since it will be this way for outgoing calls as well. 3: span 3 (not the port) 211123456: CLID, probably subject to filtering (see

Re: [asterisk-users] incoming DAHDI Channel explained

Yes, my assumption was wrong and to make things worse, my CDR data clearly show that i cannot denote incoming calls. Maybe it's time that I learn the rules as well: Analog channels do not seem to have a special identifier. The 1st call for analog channel 13 would be s.th. like DAHDI/13-1.

Re: [asterisk-users] incoming DAHDI Channel explained

B.H. Hi! On Wed, Jun 5, 2013 at 7:26 PM, jg webaccou...@jgoettgens.de wrote: For a BRI device a single span has 2 channels, a PRI device up to 30. As far as channel variables go the actual channel does not seem to get reported, but this is not really necessary. AFAIK, at least for AMI

[asterisk-users] Incoming Fax to Recipient using OCR

I have fax working but since most people and services don't know how to Fax to Extensions, I installed tesseract to convert the Fax to Text. I really only need the First Page converted and will tell Faxers to make sure they put To: Name on the cover page. tesseract is converting the entire

[asterisk-users] Incoming Fax to Recipient using OCR

I have fax working but since most people and services don't know how to Fax to Extensions, I installed tesseract to convert the Fax to Text. I really only need the First Page converted and will tell Faxers to make sure they put To: Name on the cover page. tesseract is converting the entire

Re: [asterisk-users] Incoming Fax to Recipient using OCR

To: Asterisk Users Subject: [asterisk-users] Incoming Fax to Recipient using OCR I have fax working but since most people and services don't know how to Fax to Extensions, I installed tesseract to convert the Fax to Text. I really only need the First Page converted and will tell Faxers to make sure

Re: [asterisk-users] Incoming Fax to Recipient using OCR

On Tue, Nov 6, 2012 at 1:50 PM, Roy Abshire r...@coopvr.com wrote: I have fax working but since most people and services don't know how to Fax to Extensions, I installed tesseract to convert the Fax to Text. I really only need the First Page converted and will tell Faxers to make sure they

[asterisk-users] Incoming SIP call is rejected always.

Hi Have an asterisk. Setup a couple of friends. Sip.conf - http://pastebin.com/zUgiYbBi Trying to make incoming call, and have such error(cli output) http://pastebin.com/zFfgYcNR NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from 'RMT20' (192.168.8.1:5062) to extension '4001020'

Re: [asterisk-users] Incoming SIP call is rejected always.

: [asterisk-users] Incoming SIP call is rejected always. Hi Have an asterisk. Setup a couple of friends. Sip.conf - http://pastebin.com/zUgiYbBi Trying to make incoming call, and have such error(cli output) http://pastebin.com/zFfgYcNR NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from 'RMT20

Re: [asterisk-users] Incoming SIP call is rejected always.

2012/4/17 Danny Nicholas da...@debsinc.com: Maybe it needs to be _4001020? Not, it doesn't. Actually I have traced this incoming call step by step. Real reason it refuses - wrong domain. But why it wrong - have not any idea. --

Re: [asterisk-users] Incoming SIP call is rejected always.

: Tuesday, April 17, 2012 4:58:14 PM Subject: Re: [asterisk-users] Incoming SIP call is rejected always. 2012/4/17 Danny Nicholas da...@debsinc.com: Maybe it needs to be _4001020? Not, it doesn't. Actually I have traced this incoming call step by step. Real reason it refuses - wrong domain

Re: [asterisk-users] Incoming SIP call is rejected always.

2012/4/18 Matthew Jordan mjor...@digium.com: I imagine that this is the case, as ASTERISK-19601 noted that when this situation occurs, the NOTICE message indicates that there is a failure to match the extension, as opposed to a failure to match an allowed domain. Yes, it was hell to detect

Re: [asterisk-users] Incoming SIP call is rejected always.

- Original Message - From: Yaroslav Panych panyc...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2012 6:56:17 PM Subject: Re: [asterisk-users] Incoming SIP call is rejected always. 2012/4/18 Matthew

[asterisk-users] Incoming Call Recording

Longtime lurker, first time poster. :) A client of mine is in need of having Asterisk record every call that comes in from a specific incoming route. I've added the following lines to the sip_additional.conf file, but no recordings are showing up in the /var/spool/asterisk/monitor/ folder.

Re: [asterisk-users] Incoming Call Recording

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Hall Sent: Friday, June 10, 2011 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Incoming Call Recording Longtime lurker, first

[asterisk-users] Incoming SRTP call not working with Bria iPhone Edition

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Everybody, I am experiencing some troubles with my Bria iPhone Edition (v. 1.2.8 build 5312, on iOS 4.2.1 iPhone 3G) and Asterisk 1.8.3.2 + TLS/SRTP on LAN (without NAT). With 2 computer clients (Blink, one on Mac, one on Windows/Linux),9i can

[asterisk-users] incoming

Is it possible to have Calls incoming to different DIDs? I want an AA that handles 100s of businesses. [Incoming-pizza] Exten = 4045551212,1,Goto(pizza,s,1) [Incoming-hvac] Exten = 8085551212,1,Goto(hvac,s,1) [Incoming-gutter] Exten = 6175551212,1,Goto(gutter,s,1) --

Re: [asterisk-users] incoming

Yes, I don't see why not. You just need to setup an IVR for each business and then assign each individual DID to the appropriate IVR. This may help: http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu Cheers! Rick -- Rick Hall Senior Vice President ReadyWire Multimedia Solutions

Re: [asterisk-users] incoming

Cool. So, one Asterisk machine handling up to 100 DID numbers, correct? Yes. I will have unique IVR flows/plans for each. I assume that the DID mumbers dialed would be the exaxt match needed to start the respective context. Correct? On 1/3/11, Rick Hall r...@readywire.com wrote: Yes, I don't

Re: [asterisk-users] incoming

On Sun, 2 Jan 2011, Thomas Perron wrote: Is it possible to have Calls incoming to different DIDs? Yes*, depending on whether your provider 'provides' the DID in the call setup. *) Better subjects attract more readers. More detail yields better answers. -- Thanks in advance,

Re: [asterisk-users] incoming

On Mon, Jan 03, 2011 at 02:41:36AM +0400, Thomas Perron wrote: Cool. So, one Asterisk machine handling up to 100 DID numbers, correct? As many as you like, modulo memory and CPU requirements. I assume that the DID mumbers dialed would be the exaxt match needed to start the respective context.

Re: [asterisk-users] incoming

On Mon, 3 Jan 2011, Thomas Perron wrote: So, one Asterisk machine handling up to 100 DID numbers, correct? The number of DIDs is not limited. You could handle a bazillion DIDs with a simple dial plan like: exten = _!.,1, verbose(1,[${ext...@${context}])

Re: [asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

Just out of curiosity, what country are you in? I agree with the others in this thread, this seems very bizzare that the telco requires you to do SS7 for dialup connections. I would ask them for specifics about the legal issues with what you are doing - it sounds to me like they are just trying

Re: [asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

Matt, We are located on Costa Rica and so far there's just 1 TELCO running the industrym with the CAFTA treatment the carrier had to open for interconnection but they get to define the ground rules for the interconnection. They are arguing ISDN is and end customer circuit and you cannot use it

[asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted

Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

To: asterisk-users@lists.digium.com Subject: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible?? Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem

Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

:* [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible?? Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out

Re: [asterisk-users] Incoming calls through SS7 for datamodemtransmissions - possible??

. Cary -- *From:* José Pablo Méndez Soto [mailto:aux...@gmail.com] *Sent:* Wednesday, November 24, 2010 8:34 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* ca...@usawide.net *Subject:* Re: [asterisk-users] Incoming calls through SS7

[asterisk-users] Incoming calls

Hi all, I'd like that each analog trunk of my TDM410p was received in different extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a different context and in my extensions.conf, under [default] I put such contexts and an especific estension to answer it. therefore,

Re: [asterisk-users] Incoming calls

On Thu, 18 Nov 2010, Flavio Miranda wrote: I'd like that each analog trunk of my TDM410p was received in different extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a different context and in my extensions.conf, under [default] I put such contexts and an

Re: [asterisk-users] Incoming calls

MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Thu, 18 Nov 2010 11:53:26 -0800 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incoming calls On Thu, 18 Nov 2010, Flavio Miranda wrote: I'd like that each analog trunk of my TDM410p

Re: [asterisk-users] Incoming calls

On Thu, 18 Nov 2010, Flavio Miranda wrote: Looking to dahdi show channles , I realized  that all the trunks was in the same context. So, I have changed  this and everything works! That's why I prefer to work from what Asterisk parsed the file as, not what the poster thinks :) -- Thanks in

[asterisk-users] Incoming calls

Hi all, After a lot of trouble with a TE110p working with mfcr2 , brazil variant, everything looks great,but I can not go out of my calls. When I try I receive the following log: == Using SIP RTP CoS mark 5-- Executing [33220...@local:1] Dial(SIP/4804-001a, DAHDI/g11/33220567,,T)

[asterisk-users] incoming call FXO

Hi all, Recently I have instaled one Digium TDM410 on my Asterisk. After instaled , I can do outgoing calls but I cant receive calls. I receive the following messages: chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)...[Sep

Re: [asterisk-users] incoming call FXO

On 09/15/2010 07:20 AM, Flavio Miranda wrote: Recently I have instaled one Digium TDM410 on my Asterisk. After instaled , I can do outgoing calls but I cant receive calls. I receive the following messages: chan_dahdi.c: Got event 2 (Ring/Answered)... [Sep 14 11:24:44] NOTICE[2654]

Re: [asterisk-users] incoming call FXO

As Kevin said, you need to define an 's' extension where the calls will be answered. Seems like you are using default configuration. Open file 'extensions.conf' in /etc/asterisk folder and look for context named [default]. If it is not there, create one and add something under it, e.g., [default]

Re: [asterisk-users] incoming call FXO

Ok. Problem solved . Thank you very much!!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Wed, 15 Sep 2010 09:56:36 -0400 From: zisha...@gmail.com To: kpflem...@digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] incoming

[asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' =1. Wait(1)

Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

On Wed, Aug 4, 2010 at 8:52 PM, Joe Wood sch...@gmail.com wrote: Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [

Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com wrote: You don't have any extensions in your default context that match the extension that your sip peer is dialing in on.  's' is not a default extension for SIP...try using _X., and see what you get.  Bump up

Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote: I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com wrote: You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a

Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote: On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote: I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com wrote: You don't have any extensions in your default context that

Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

On Wed, Aug 4, 2010 at 10:25 PM, Joe Wood sch...@gmail.com wrote: On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote: On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote: My experience with Asterisk in the past has been with inbound analog lines so that

Re: [asterisk-users] Incoming call doesn't finish when internal phone hangs up

: [asterisk-users] Incoming call doesn't finish when internal phone hangs up To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: aanlktikxafnxhbsws0ov4u5ht3yjbeevuh26vehrg...@mail.gmail.com Content-Type: text/plain; charset=ISO

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