Re: [asterisk-users] Asterisk load balancing for TCP/SIP and RTP?
Have a look at this page for HA ideas: http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Design There are a lot of tradeoffs in design, and easy to confuse load balancing with HA From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tickling Contest Sent: Tuesday, March 29, 2016 7:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk load balancing for TCP/SIP and RTP? Has anyone fronted Asterisk with HAProxy? If so, what is a good production configuration for Asterisk? I need direct_media=yes (and so I have to LB RTP to the same Asterisk server). If HAProxy is not a good solution, what other solutions do you propose? Any insight appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk load balancing for TCP/SIP and RTP?
When using HA you have one to one. We use OpenSipS to load balance so you never lose more than half your calls. Regards, Dovid -Original Message- From: Tickling Contest Sender: asterisk-users-bounces@lists.digium.comDate: Tue, 29 Mar 2016 19:02:00 To: Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk load balancing for TCP/SIP and RTP? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk load balancing for TCP/SIP and RTP?
Has anyone fronted Asterisk with HAProxy? If so, what is a good production configuration for Asterisk? I need direct_media=yes (and so I have to LB RTP to the same Asterisk server). If HAProxy is not a good solution, what other solutions do you propose? Any insight appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13.8.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New Features made in this release: --- * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write contents to file (Reported by Ray Crumrine) * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel Journo) * ASTERISK-25480 - [patch]Add field PauseReason on QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena) Bugs fixed in this release: --- * ASTERISK-25849 - chan_pjsip: transfers with direct media sometimes drops audio (Reported by Kevin Harwell) * ASTERISK-25113 - install_prereq in Debian 8 without "standard system utilities" (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so (Reported by Sergio Medina Toledo) * ASTERISK-25023 - Deadlock in chan_sip in update_provisional_keepalive (Reported by Arnd Schmitter) * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local channel (Reported by Filip Frank) * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when separating multiple AORs (Reported by Mateusz Kowalski) * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into Stasis application. (Reported by Javier Riveros ) * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean Bright) * ASTERISK-25582 - Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt Jordan) * ASTERISK-25811 - Unable to delete object from sorcery cache (Reported by Ross Beer) * ASTERISK-25800 - [patch] Calculate talktime when is first call answered (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to PJSIP requirement (Reported by Gergely Dömsödi) * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub (Reported by Jacques Peacock) * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing OutboundSubscriptionDetail ami action (Reported by Kevin Harwell) * ASTERISK-25721 - [patch] res_phoneprov: memory leak and heap-use-after-free (Reported by Badalian Vyacheslav) * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes returns garbage (Reported by Etienne Lessard) * ASTERISK-25751 - res_pjsip: Support pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp) * ASTERISK-25606 - Core dump when using transports in sorcery (Reported by Martin MouÄka) * ASTERISK-20987 - non-admin users, who join muted conference are not being muted (Reported by hristo) * ASTERISK-25737 - res_pjsip_outbound_registration: line option not in Alembic (Reported by Joshua Colp) * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in udptl_rx_packet cause ast_frdup crash (Reported by Walter Doekes) * ASTERISK-25742 - Secondary IFP Packets can result in accessing uninitialized pointers and a crash (Reported by Torrey Searle) * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST Vulnerability - Investigate vulnerability of HTTP server (Reported by Alex A. Welzl) * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with non-default timert1 (Reported by Alexander Traud) * ASTERISK-25702 - PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by Nic Colledge) * ASTERISK-25730 - build: make uninstall after make distclean tries to remove root (Reported by George Joseph) * ASTERISK-25725 - core: Incorrect XML documentation may result in weird behavior (Reported by Joshua Colp) * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in sip_sipredirect (Reported by Badalian Vyacheslav) * ASTERISK-25709 - ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Reported by Mark Michelson) * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported by Badalian Vyacheslav) * ASTERISK-25685 - infrastructure: Run alembic in Jenkins build script (Reported by Joshua Colp) * ASTERISK-25712 - Second call to already-on-call phone and Asterisk sends "Ready" (Reported by Richard Mudgett) * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow (Reported by Badalian Vyacheslav) * ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report incorrect values (Reported by Gianluca Merlo) * ASTERISK-25611 - core: threadpool thread_timeout_thrash unit test sporadically failing (Reported by Joshua Co
[asterisk-users] Asterisk 11.22.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.22.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.22.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: --- * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to data corruption (Reported by Gianluca Merlo) * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local channel (Reported by Filip Frank) * ASTERISK-25800 - [patch] Calculate talktime when is first call answered (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes returns garbage (Reported by Etienne Lessard) * ASTERISK-20987 - non-admin users, who join muted conference are not being muted (Reported by hristo) * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST Vulnerability - Investigate vulnerability of HTTP server (Reported by Alex A. Welzl) * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in udptl_rx_packet cause ast_frdup crash (Reported by Walter Doekes) * ASTERISK-25742 - Secondary IFP Packets can result in accessing uninitialized pointers and a crash (Reported by Torrey Searle) * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with non-default timert1 (Reported by Alexander Traud) * ASTERISK-25730 - build: make uninstall after make distclean tries to remove root (Reported by George Joseph) * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in sip_sipredirect (Reported by Badalian Vyacheslav) * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported by Badalian Vyacheslav) * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow (Reported by Badalian Vyacheslav) * ASTERISK-25701 - core: Endless loop in "core show taskprocessors" (Reported by ibercom) * ASTERISK-25700 - main/config: Clean config maps on shutdown. (Reported by Corey Farrell) * ASTERISK-25690 - Hanging up when executing connected line sub does not cause hangup (Reported by Joshua Colp) * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh reload' cause a crash (Reported by Sean Bright) * ASTERISK-25394 - pbx: Incorrect device and presence state when changing hint details (Reported by Joshua Colp) * ASTERISK-25640 - pbx: Deadlock on features reload and state change hint. (Reported by Krzysztof Trempala) * ASTERISK-25681 - devicestate: Engine thread is not shut down (Reported by Corey Farrell) * ASTERISK-25680 - manager: manager_channelvars is not cleaned at shutdown (Reported by Corey Farrell) * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by Corey Farrell) * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported by Corey Farrell) * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey Farrell) * ASTERISK-25647 - bug of cel_radius.c: wrong point of ADD_VENDOR_CODE (Reported by Aaron An) * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade Brandon) * ASTERISK-25442 - using realtime (mysql) queue members are never updated in wait_our_turn function (app_queue.c) (Reported by Carlos Oliva) * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by sungtae kim) Improvements made in this release: --- * ASTERISK-24813 - asterisk.c: #if statement in listener() confuses code folding editors (Reported by Corey Farrell) * ASTERISK-25767 - [patch] Add check to configure for sanitizes (Reported by Badalian Vyacheslav) * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the core set (Reported by Rusty Newton) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI error "ROSE REJECT"
Perhaps it's taking a bit longer in the network for the media path to
open after the CONNECT, which would explain why the first digit is not
being detected.
Also, what changed last week?
I don't see the ROSE REJECT message anywhere in the pri debug -
perhaps you didn't catch it.
Matthew Fredrickson
On Fri, Mar 25, 2016 at 9:15 PM, Carlos Chavez wrote:
> On 2016-03-25 16:02, Matt Fredrickson wrote:
>>
>> PRI debug of the entire call would be great, also, switchtype would be
>> awesome as well.
>>
>> Thanks!
>>
>> Matthew Fredrickson
>>
>> On Thu, Mar 24, 2016 at 4:07 PM, Carlos Rojas wrote:
>>>
>>> Hi
>>>
>>> Did you activate the pri debug on the cli asterisk?
>>>
>>> On Thu, Mar 24, 2016 at 12:59 PM, Carlos Chavez
>>> wrote:
We've been having some problems with an E1 PRI line for a few days. We
get the following errors:
[Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT:
[Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2INVOKE
ID:
316
[Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2PROBLEM:
Invoke: Unrecognized Operation
The telephone company says that everything is fine on their side,
obviously. The problems started a few days ago when a user reported
that
incoming calls get dropped when you try to dial a particular extension
from
the main IVR. We are using Asterisk 1.8.15-cert2 on a CentOS 6.7
server,
DAHDI 2.6.1 and libpri 1.4. Any recommendations?
--
>
>
> system.conf:
> # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
> span=1,2,0,cas,hdb3
> cas=1-15:1101
> cas=17-31:1101
>
> # Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
> span=2,1,0,ccs,hdb3
> # termtype: te
> bchan=32-46
> dchan=47
> bchan=48-62
>
> loadzone= mx
> defaultzone = mx
>
> chan_dahdi.conf:
> language=es
> context=e1-incoming
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> canpark=no
> usecallingpres=no
> callwaitingcallerid=no
> threewaycalling=no
> transfer=yes
> cancallforward=no
> callreturn=no
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=yes
> rxgain=0.0
> txgain=0.0
> accountcode=E1
> amaflags=default
> signalling=pri_cpe
> pridialplan=unknown
> prilocaldialplan=unknown
> switchtype=euroisdn
> overlapdial=no
> immediate=no
> group=2
> faxdetect=no
> callerid=asreceived
> mohinterpret=default
> mohsuggest=default
> dahdichan=32-46,48-62
>
> Here is the pri debug:
> -- Accepting call from '55' to '5732' on channel 0/1, span 2
> -- Executing [5732@e1-incoming:1] Goto("DAHDI/i2/55-3c",
> "menu-gci,s,1") in new stack
> -- Goto (menu-gci,s,1)
> -- Executing [s@menu-gci:1] Wait("DAHDI/i2/55-3c", "2") in new
> stack
> -- Executing [s@menu-gci:2] Answer("DAHDI/i2/55-3c", "") in new
> stack
> PRI Span: 2 q931.c:4683 q931_connect: Call 48 enters state 8 (Connect
> Request). Hold state: Idle
> PRI Span: 2
> PRI Span: 2 > DL-DATA request
> PRI Span: 2 > Protocol Discriminator: Q.931 (8) len=14
> PRI Span: 2 > TEI=0 Call Ref: len= 2 (reference 48/0x30) (Sent to
> originator)
> PRI Span: 2 > Message Type: CONNECT (7)
> PRI Span: 2 TEI=0 Transmitting N(S)=98, window is open V(A)=98 K=7
> PRI Span: 2
> PRI Span: 2 > Protocol Discriminator: Q.931 (8) len=14
> PRI Span: 2 > TEI=0 Call Ref: len= 2 (reference 48/0x30) (Sent to
> originator)
> PRI Span: 2 > Message Type: CONNECT (7)
> PRI Span: 2 > [18 03 a9 83 81]
> PRI Span: 2 > Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI)
> Spare: 0 Exclusive Dchan: 0
> PRI Span: 2 > ChanSel: As indicated in following
> octets
> PRI Span: 2 > Ext: 1 Coding: 0 Number Specified
> Channel Type: 3
> PRI Span: 2 > Ext: 1 Channel: 1 Type: CPE]
> PRI Span: 2 > [1e 02 81 82]
> PRI Span: 2 > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU)
> standard (0) 0: 0 Location: Private network serving the local user (1)
> PRI Span: 2 > Ext: 1 Progress Description:
> Called equipment is non-ISDN. (2) ]
> -- Executing [s@menu-gci:3] BackGround("DAHDI/i2/55-3c",
> "menugci") in new stack
> -- Playing 'menugci.slin' (language 'es')
> PRI Span: 2
> PRI Span: 2 < Protocol Discriminator: Q.931 (8) len=5
> PRI Span: 2 < TEI=0 Call Ref: len= 2 (reference 48/0x30) (Sent from
> originator)
> PRI Span: 2 < Message Type: CONNECT ACKNOWLEDGE (15)
> PRI Span: 2 Received message for call 0x2c5e9c10 on 0x326b290 TEI/SAPI
> 0/0, call->pri is 0x326b290 TEI/SAPI 0/0
> PRI Span: 2 q931.c:7024 post_handle_q931_message: Call 48 enters state 10
> (Active). Hold state: Idle
> [Mar 25 20:01:52] WARNING[14859]: pbx.c:5465 __ast_pbx_run: Invalid
> extension '8', but no rule 'i' or 'e' in context 'menu-gci'
> PRI Span: 2 q931_hangup: other hangup
> PRI Span: 2 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active,
> peerstate Active, hold-state Idle
> PRI Spa
Re: [asterisk-users] SIP trunk with whatsapp
On 03/29/16 17:03, Vitor Mazuco wrote: > Is possible with Telegram? Telegram does not support voice calls for humans either. It's strictly an IM system. They do have a bot API if you want to interface some system with their messaging system. With it you can send text and also pictures, recordings and other datatypes: https://core.telegram.org/bots Not sure if this is what you are looking for. -- Guido Falsi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Client TLS certificates for auth ?
But what is the problem even if somehow your password will be stolen hacker can't make a call because he needs certificate.of course if U setup ext to use TLS only On Mar 29, 2016 5:32 PM, "Markos Vakondios" wrote: > This would be very interesting, as we could register SIP devices securely > over the internet without the need for VPN. > Asterisk of course must accept only trusted client certificates the same > way an OpenVPN server does. > Anyone operating his/her remote endpoints like this? > Anyone advising against this solution? > > On 29 March 2016 at 04:51, Kevin Long wrote: > >> >> >> I use TLS and SRTP on my Asterisk servers. The server certificates are >> signed by my internal CA, and the Root CA cert is distributed to the phones >> and soft phones so they will trust the server without warning. >> >> It is not clear to me if Asterisk can be configured to actually reject >> client connections/registrations from peers which do not possess a client >> certificate which has been signed by a particular CA ? >> >> If so, could it be such that the common name in the client certificate >> would need to match the username or Asterisk “extension” ? >> >> >> I’m wondering if this can be done , to have a second factor of >> authentication besides the SIP secret , since in my current setup, despite >> using a TLS/SSL cert for the server, the server only verifies the client by >> the SIP secret. >> >> Regards, >> >> Kevin Long >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk with whatsapp
Is possible with Telegram? 2016-03-29 9:39 GMT-03:00, Emiliano Vazquez : > El 29/03/16 a las 08:29, Steve Howes escribió: >> I don't think you can. Whatsapp is a closed system. >> >> Steve > And they change your code every day and make it always obfuscated. > > https://github.com/tgalal/yowsup/issues/887 > > Best regards. > > Emiliano. > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Client TLS certificates for auth ?
This would be very interesting, as we could register SIP devices securely over the internet without the need for VPN. Asterisk of course must accept only trusted client certificates the same way an OpenVPN server does. Anyone operating his/her remote endpoints like this? Anyone advising against this solution? On 29 March 2016 at 04:51, Kevin Long wrote: > > > I use TLS and SRTP on my Asterisk servers. The server certificates are > signed by my internal CA, and the Root CA cert is distributed to the phones > and soft phones so they will trust the server without warning. > > It is not clear to me if Asterisk can be configured to actually reject > client connections/registrations from peers which do not possess a client > certificate which has been signed by a particular CA ? > > If so, could it be such that the common name in the client certificate > would need to match the username or Asterisk “extension” ? > > > I’m wondering if this can be done , to have a second factor of > authentication besides the SIP secret , since in my current setup, despite > using a TLS/SSL cert for the server, the server only verifies the client by > the SIP secret. > > Regards, > > Kevin Long > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk with whatsapp
El 29/03/16 a las 08:29, Steve Howes escribió: I don't think you can. Whatsapp is a closed system. Steve And they change your code every day and make it always obfuscated. https://github.com/tgalal/yowsup/issues/887 Best regards. Emiliano. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk with whatsapp
On 28/03/16 12:46, bilal ghayyad wrote: Does anyone has information if possible to setup SIP trunk with whatsapp? How can we let asterisk send and receive calls from whatsapp? I don't think you can. Whatsapp is a closed system. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone Number Validation
On Tuesday 29 Mar 2016, Rizwan H Qureshi wrote: > Hi Everyone, > I need to develop a service which tells me whether a given phone number is > in service and is valid or not. It can be international number. This is > basically to clean the list of leads we have. Is there any service which > can give me the required information? > > I currently have an international numbering plan database which only tells > me if the given phone number is in valid format up to a certain area code. > But I need to know whether it will ring or not. Any help will be > appreciated. There is exactly one way to find out whether or not a given telephone number will answer: Dial it. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone Number Validation
On Tue, 29 Mar 2016 09:53:15 +0100 Rizwan H Qureshi wrote: > I need to develop a service which tells me whether a given phone > number is in service and is valid or not. It can be international > number. This is basically to clean the list of leads we have. Is > there any service which can give me the required information? Sounds like a service that spammers have been searching for since Alex called Tommy and asked him to "Come here." Such a service is impossible. There is no way to tell if a number will be answered without actually calling it. Even then you don't know whether an out of service is temporary or permanent. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone Number Validation
On Tue, 29 Mar 2016 09:53:15 +0100 Rizwan H Qureshi wrote: > Hi Everyone, > I need to develop a service which tells me whether a given phone number is > in service and is valid or not. It can be international number. This is > basically to clean the list of leads we have. Is there any service which > can give me the required information? > > I currently have an international numbering plan database which only tells > me if the given phone number is in valid format up to a certain area code. > But I need to know whether it will ring or not. Any help will be > appreciated. > > Thanks > > Best Ragards > Rizwan H Qureshi > > V: +44 (0) 7544180726 > linkedin.com/in/rhqureshi This might come handy: https://github.com/googlei18n/libphonenumber Regards, Lefteris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone Number Validation
Hi Everyone, I need to develop a service which tells me whether a given phone number is in service and is valid or not. It can be international number. This is basically to clean the list of leads we have. Is there any service which can give me the required information? I currently have an international numbering plan database which only tells me if the given phone number is in valid format up to a certain area code. But I need to know whether it will ring or not. Any help will be appreciated. Hi! I am doing something similar. Country codes are available from ITU-T. Country codes are available for every country, except for the North American Numbering Plan, which covers essentially North America. NANP numbers have a simple structure (with little oddities), which is not generally valid outside their domain, so it is difficult to check the validity of numbers (unless you are willing to work through the regulations of every country you want to cover). For example, a complete German phone number, including the equivalent of NPA and NXX, can be between 5 and 15 digits. The system is (almost) strictly hierarchical, but requires detailed knowledge, i.e. you do need an algorithm that figures out the area code. There are also separate number ranges for mobile phone numbers. In practice there can be more than 15 numbers, depending on the country, and whether the regulators are not particularly strict in enforcing a specific length of phone numbers (for ISDN lines). Generally, you cannot know whether dialing a number will ring the other end, or not. If all channels are already occupied for a T1 or E1 connection, the last exchange station will already signal unavailability, i.e. "user busy" may be signaled by the user or the network. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phone Number Validation
Hi Everyone, I need to develop a service which tells me whether a given phone number is in service and is valid or not. It can be international number. This is basically to clean the list of leads we have. Is there any service which can give me the required information? I currently have an international numbering plan database which only tells me if the given phone number is in valid format up to a certain area code. But I need to know whether it will ring or not. Any help will be appreciated. Thanks Best Ragards Rizwan H Qureshi V: +44 (0) 7544180726 linkedin.com/in/rhqureshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
