Re: [asterisk-users] Paging systems?
Michael Based on how I read the manual if you connect the TIP and RING of the ATA to the right pairs you should be able to send a call to the paging box. It looks that when the page call is picked up by the paging system you would then press a zone 1-9 or 0 for all. The page would then bridge to the desired zone. the page would complete when the call is hung up. You would likely need to make sure the ATA is using current loop disconnect or reverse to ensure hang-up. I think it should be the PABX config using the Figure 3 configuration. Best of luck Bryant Zimmerman Sr. Systems Architect Grand Dial Communications, A ZK Tech Inc. Company 616-299-5607 (mobile) 616-855-1030 Ext. 2003 (office) From: Michael Munger Sent: 3/21/19 7:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion , John Novack Subject: Re: [asterisk-users] Paging systems? Excellent point. This is it: https://www.valcom.com/pdf/v-1109rthf.pdf Get Outlook for Android On Thu, Mar 21, 2019 at 7:22 PM -0400, "John Novack" wrote: Michael Munger wrote: Does anyone have an (overhead) paging system that they like that works with SIP? We’ve got a client with an old paging system that (supposedly) just takes an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t auto-answer the call, so paging never happens. Does it expect to see a POTS line with battery on it? Then a Cisco or other ATA that would work to supply service to a POTS phone should work OR: Does it expect to see a POTS connection from a PBX trunk, and supply battery TO the trunk? Then you would need a Cisco or other ATA with an FXO connection. Both types of paging systems have been made and both styles of connections have existed through the last 30 + years, and since you haven't revealed the brand and model of paging system it makes troubleshooting difficult. Using the existing system can be made to work I use a very old Harris PagePak VS that was used with a Western Electric Horizon system back in the dark ages with Asterisk John Novack -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging systems?
Michael We use Bogan UTI1 box in conjunction with an ATA to patch to any overhead paging system. You patch the box directly into the amp line in for the overhead. When you call the extension it answers and puts the audio on the line to the PA. If you only have a few speakers the UTI1 can even handle being the amp for a few speakers. Bryant Zimmerman Sr. Systems Architect Grand Dial Communications, A ZK Tech Inc. Company 616-299-5607 (mobile) 616-855-1030 Ext. 2003 (office) From: Darryl Moore Sent: 3/21/19 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging systems? For a paging system? No you don't. A number of SNOM PA1's and a few grandstream phones and you're golden. If you do need FXO or FXS, they are just as easy to setup as well, and there are lots to choose from. On Thu, Mar 21, 2019, 4:45 PM Ryan, Travis, wrote: You need more than an ATA. You need something with an FSO and FXO. I’ve used Linksys/SPA3102-3.3.6 and been happy with it. From: asterisk-users On Behalf Of Sebastian Nielsen Sent: Thursday, March 21, 2019 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging systems? How did the page system answer the call when it was used with the analog system? You could propably ”fake” those signals from inside asterisk, and cause it to answer. Från: asterisk-users För Michael Munger Skickat: den 21 mars 2019 20:00 Till: asterisk-users@lists.digium.com Ämne: [asterisk-users] Paging systems? Does anyone have an (overhead) paging system that they like that works with SIP? We’ve got a client with an old paging system that (supposedly) just takes an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t auto-answer the call, so paging never happens. Michael J. Munger, dCAP, MCPS, MCNPS, MBSS Microsoft Certified Professional Microsoft Certified Small Business Specialist Digium Certified Asterisk Professional High Powered Help, Inc. p: 678-905-8569 w: hph.io e: m...@hph.io -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13.18.4 - New Error PJLIB_UTIL_EDNS_REFUSED
We just updated from 13.17.1 to 13.18.4 and are noticing a new error [2017-12-21 10:12:48] ERROR[32343]: res_pjsip.c:3850 endpt_send_request: Error 320055 'DNS "Refused" (PJLIB_UTIL_EDNS_REFUSED)' sending OPTIONS request to endpoint 6162480909.8009 The DNS on the system seems to be working find. Anyone have an idea what could be triggering this issue? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] user-agent access from pjsip
I am trying to get the user-agent from extensions registered via pjsip. With sip we could do a sip show peer peername and it would list the user-agent string. In a pjsip deployment it looks like this info is likely in the contact. I know we can access it from the dialplan, but this is only works when a call occurs. How can we get the user-agent for extensions from the console. We need this for firmware version checking of extensions as many providers include that in the user-agent. Any ideas as the pjsip show contact contactname does not return any real helpful info to the command line. Please advise if you are able. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP Asteirks 13 - Audio Jitter in one direction only
?We have upgraded a system from Asterisk 11 to Asterisk 13 with pjsip. We are experiencing random Jitter on outbound calls. This was not occurring when running asterisk 11. We have two IP's bound to pjsip one on the private vlan network the phones are on and the asterisk one on the asterisk wan vlan. We record the calls on the asterisk switch so we have the call legs. It appears that the audio is making it to the switch fine, but is being garbled before it leaves asterisk to the destination carrier. We have all media running through the server and this is happening when there is only 1 to 2 calls on the line. The cpu, and memory are not even being pushed. We are running G711 as the codec so there should be no real transcoding occurring.. What could be causing this. The users are very upset. This is a very transient issue so the breakup is can occur for two to four seconds and then goes away. It is like asterisk and pjsip are screwing with the audio. Please advise. zktech -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP add header not working
Andre
For this to work we have had to go to using the b() option in the dial
legs for the calls that are pasting up.
You call a context that gets run before the calls are made on each
channel. This allows you to add headers to the new pjsip channels.
It works well. You can also set variables with the _ option to trigger
which headers you want to add..
The example below would add "ThisHeader", "ThatHeader" and "Call-Info" to
the new channel created in the dial. You could use combinations of other
variables and augment these methods to meet almost any need.
Exp
[OutboundDial]
exten => _XX,1,NoOp(Dial Exp)
exten => _XX,n,Set(_var1setinparrent=1) ;;Set Variable so that
when you call the b() option context in your dial the first header is
added
exten => _XX,n,Set(_var2setinparrent=1) ;;Set Variable so that
when you call the b() option context in your dial the second header is
added
exten => _XX,n,Set(_varAddSessionInparrent=1) ;;Set Variable so
that when you call the b() option context in your dial the second header is
added
exten =>
_XX,n,Dial(pjsip/333222@vendortrunk,b(AddpjsipHeaders^s^1))
[AddpjsipHeaders]
exten =>s,1,Gosubif({"$[var1setinparrent}}"="1"]?ThisHeader,1)
exten =>s,n,Gosubif({"$[var2setinparrent}}"="1"]?ThatHeader,1)
exten
=>s,n,Gosubif({"$[varAddSessionInparrent}}"="1"]?addSessionCallInfo,1)
exten => ThisHeader,1,Set(PJSIP_HEADER(add,ThisHeader)=ValueToSet)
exten => ThisHeader,n,Return()
exten => ThatHeader,1,Set(PJSIP_HEADER(add,ThatHeader)=ValuetoSet)
exten => ThatHeader,n,Return()
exten =>
addSessionCallInfo,1,Set(PJSIP_HEADER(add,Call-Info)=\;answ
er-after=0)
exten => addSessionCallInfo,n,Return()
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: "Andre Gronwald"
Sent: Monday, October 2, 2017 11:07 AM
To: "asterisk-users"
Subject: [asterisk-users] PJSIP add header not working
Hi,
I am trying to add a custom header to my calls to map several call-legs
into a global call for viewing.
For this to work I read the call-id from pjsip-channel and write it into
X-CID:
##
-- Executing [s@macro-dialout-trunk-predial-hook:4]
Set("PJSIP/10-0006",
"pjsipCallId=313530363933383438363436353930-1gh0bjceo933") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:5]
Set("PJSIP/10-0006",
"PJSIP_HEADER(add,X-CID)=313530363933383438363436353930-1gh0bjceo933") in
new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/10-0006",
"0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/10-0006",
"1?Set(CONNECTEDLINE(num,i)=0xx)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/10-0006",
"1?Set(CONNECTEDLINE(name,i)=CID:3x)") in new stack
-- Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/10-0006",
"0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)3x)") in new stack
-- Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/10-0006",
"0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:23] Dial("PJSIP/10-0006",
"PJSIP/0xx@3x,300,T") in new stack
-- Called PJSIP/0xx@3x
<--- Transmitting SIP request (991 bytes) to UDP:217.23.24.100:5060 --->
INVITE sip:0xxx...@sip.provid.er:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.253.185:15070;rport;branch=z9hG4bKPj453d15e0-de58-4945-8b95-d05b16b9
e4c3
From:
;tag=080788ac-7c10-4cf3-86b3-359764ffb5a2
To:
Contact:
Call-ID: de41b93b-51d8-44b5-9c34-f2c0928192b0
CSeq: 1519 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-14.0.1.10(14.6.2)
Content-Type: application/sdp
Content-Length: 308
v=0
o=- 1719768133 1719768133 IN IP4 192.168.253.185
s=Asterisk
c=IN IP4 192.168.253.185
t=0 0
m=audio 55112 RTP/AVP 107 9 8 3 101
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
<--- Received SIP response (559 bytes) from UDP:217.23.24.100:5060 --->
[...]
##
But I can't see that header anywhere in my call-legs. What am I missing?
kind regards,
andre
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[asterisk-users] Bug in func_odbc module
Hey all I have code we are moving from an early asterisk 13 system to the latest build. The issue we are having is func_odbc calls are acting incorrectly. We have tables that have fields with null values in them. On the new system when we read a field with a null value it is copying the value from the previous filed into the value and not leaving the filed as null or blank string. So what is happening is we get variables inside of our dialplan that have values from other variables fields. As soon as the system hits a value with a non null field, even a filed with an empty string it self corrects until it hits another field value with a null string in it. Any thoughts on how and when this could get fixed? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk pjsip registration issues - Solved
Dave from_user fixed the issue. Thank You Thank You Thank You I was about ready to chuck pjsip. The lack of good / complete documentation is a real problem. Man you saved me another late night. Thanks Bryant From: "Dave Platt" Sent: Tuesday, September 26, 2017 3:28 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk pjsip registration issues > Hey all > > I am trying to register a PJSIP server on our office to an Asterisk 11 > chan_sip server in a datacenter. > > I keep getting > WARNING[18084]: res_pjsip_outbound_authenticator_digest.c:178 > digest_create_request_with_auth_from_old: Host: 'XXX.XXX.XXX.XXX:5060': > Unable to create request with auth. No auth credentials for realm(s) > 'asterisk' in challenge. > > Any insights would be appreciated I have been banging my head for several > days now. I ran into a very similar problem when I tried to switch my PJSIP service with Vitelity from "fixed IP address" to "registration-based". I would try to place a call, and it would simply time out and then get a "busy here" error from Vitelity. Calls to a similar Vitelity sub-account from a Zoiper soft-phone worked just fine. I wiresharked the sessions and found that the critical difference seemed to be in the From: and Contact: headers. Zoiper set these to the Vitelity sub-account name (the registration name) while PJSIP just set them to "asterisk". I checked the PJSIP wizard file, and found that the outbound authentication object had the right username information in it, so that wasn't the problem. After stumbling around for hours, I found that it's necessary to set the "from_user" parameter in the endpoint object to match the username in the outbound authentication object. This causes PJSIP to send this value (rather than "asterisk") in the From and Contact fields of the INVITE, and this apparently gives the far end the information it needs to issue a proper credentials challenge. Once I added this one line to my definition and restarted, outbound calls worked like a charm. So, in pjsip_wizard, one would write something like [peername] type = wizard transport = transport-udp remote_hosts = outbound.peer.com sends_auth = yes endpoint/context = outbound endpoint/from_user = MYNAME outbound_auth/username = MYNAME outbound_auth/password = MYPASSWORD Modify and embellish as required. If you're writing your PJSIP objects individually rather than via the wizard, just set the fields in those objects appropriately. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk pjsip registration issues
Hey all I am hoping someone can assist I have now spent over a week trying to figure out what is going on with PJSIP registrations. I am able to register handsets against an asterisk 13 server running pjsip, but I am not able to get pjsip to register out to an older chan_sip asterisk server. If I drop the registration I can make things work, but when I have to register the asterisk - pjsip server against another server the registration completes, but I can not send any calls across the registration, nor will it handle options correctly as well. We keep getting ... No auth credentials for realm(s) 'aster...@xxx.xxx.xxx.xxx' in challenge. in one form or another, and I have been unable to find any definitive documentation on what is at cause for this. In some areas I have seen responses saying it is an issue with realms so I have tried with and without but no success. I really need some direction on this. This is the last issue I know of that is holding up us from moving to pjsip. If I can't get asterisk / pjsip to register and send authenticated messages than it can't work for replacing chan_sip in all situations. What am I doing wrong. [zktech_trunk] type=registration endpoint=zktech_trunk transport=udp-nat outbound_auth=zktech_trunk server_uri=sip:acct.8...@xxx.xxx.xxx.xxx client_uri=sip:acct.8...@xxx.xxx.xxx.xxx contact_user=zktech_trunk retry_interval=60 forbidden_retry_interval=600 expiration=3600 line=yes [zktech_trunk] type=auth auth_type=userpass password=rossi72v8qr username=ACCT.8009 realm=aster...@xxx.xxx.xxx.xxx [zktech_trunk] type=aor max_contacts=1 contact=sip:acct.8...@privxxx.xxx.xxx.xxx:5060 qualify_frequency=60 Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registering Asterisk 13 server PJSIP to Asterisk 11 SIP
Hey all I am trying to register a PJSIP server on our office to an Asterisk 11 chan_sip server in a datacenter. I keep getting WARNING[18084]: res_pjsip_outbound_authenticator_digest.c:178 digest_create_request_with_auth_from_old: Host: 'XXX.XXX.XXX.XXX:5060': Unable to create request with auth. No auth credentials for realm(s) 'asterisk' in challenge. Any insights would be appreciated I have been banging my head for several days now. Thanks bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime pjsip issues
Original Message > From: "Joshua Colp" > Sent: Friday, September 15, 2017 11:31 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Realtime pjsip issues > > On Fri, Sep 15, 2017, at 12:18 PM, Bryant Zimmerman wrote: > > Joshua > > > > We are using MariaDB as the database storage. > > We have recreated the database tables with alembic. > > > > Test 1: > > We enable tables for aors, auths and endpoints only. With cache turned > > off the end point registers successfully We have no way to get any > > feed > > back as pjsip show/list returns no objects found. pjsip send notify > > cmd > > endpoint -- does not work as it says there is no endpoint. endpoint > > can > > send a call as it appears to be registered, we have no way to confirm > > this > > form the console but calls come in. > > > > The show and list commands are supposed to work, even without caching > being enabled. Your problem is therefore at the realtime level. Calls > coming in should appear on the console, and the endpoint name will be in > the channel name. Enabling caching just masks it some because things > exist in the cache for a bit. > > > > > I can offer the following: > > A dump of the database schema that alembic is creating. > > extconfig.config > > sorcery.conf > > Feel free to provide these and me (or another individual) may pick out > what is wrong. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > I have linked to a zip file containing a dump of my sql schema (MySQL), extconfig.conf, sorcery.conf dumps.zip Hopefully someone can see what might be causing our issues with the pjsip realtime system. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime pjsip issues
Joshua We are using MariaDB as the database storage. We have recreated the database tables with alembic. Test 1: We enable tables for aors, auths and endpoints only.With cache turned off the end point registers successfullyWe have no way to get any feed back as pjsip show/list returns no objects found. pjsip send notify cmd endpoint -- does not work as it says there is no endpoint. endpoint can send a call as it appears to be registered, we have no way to confirm this form the console but calls come in. Test 2: We enable cache on the endpoints, auth and aors in the sorcery.conf endpoint/cache = memory_cache,object_lifetime_stale=600,object_lifetime_maximum=1800,expire_o n_reload=yes,full_backend_cache=yes auth/cache=memory_cache,expire_on_reload=yes aor/cache = memory_cache,object_lifetime_stale=1500,object_lifetime_maximum=1800,expire_ on_reload=yes,full_backend_cache=yes We now get an error:[2017-09-15 11:02:04] WARNING[3375]: res_pjsip_registrar.c:744 registrar_on_rx_request: AOR '6162480909-300' has no configured max_contacts. Endpoint '6162480909-300' unable to register The aors entry has the max_contacts set to 1 but the error still occurs. pjsip show/list shows the endpoint shows endpoints, aors, auths but registration fails Test 3: We enable cache on the endpoints, auth and aors in the sorcery.conf endpoint/cache = memory_cache,object_lifetime_stale=600,object_lifetime_maximum=1800,expire_o n_reload=yesauth/cache=memory_cache,expire_on_reload=yes aor/cache = memory_cache,object_lifetime_stale=1500,object_lifetime_maximum=1800,expire_ on_reload=yes Endpoint registers pjsip show/list endpoints works the first time and fails there after. UBNTU-ROSSI-GUEST*CLI> pjsip show endpoints Endpoint: I/OAuth: Aor: Contact: Transport: Identify: Match: Channel: Exten: CLCID: === === Endpoint: 6162480909-300 Not in use0 of inf InAuth: 6162480909-300/6162480909-300 Aor: 6162480909-300 1 Contact: 6162480909-300/sip:6162480909-300@192.168. 0475d46ff2 Unknown nan Transport: udp-nat udp 0 0 0.0.0.0:5060 Objects found: 1 UBNTU-ROSSI-GUEST*CLI> pjsip show endpoints No objects found. pjsip show/list shows the endpoint fails ever time after the first. Test 4: Test 1: with the addition of the contacts entry as realtime in sorcery.confWe get error on registration attempt: [2017-09-15 11:16:07] WARNING[3591]: res_config_odbc.c:120 custom_prepare: SQL Prepare failed! [INSERT INTO ps_contacts (id, via_addr, qualify_timeout, call_id, reg_server, path, endpoint, via_port, authenticate_qualify, uri, qualify_frequency, user_agent, expiration_time, outbound_proxy) VALUES (?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?)] [2017-09-15 11:16:07] ERROR[3591]: res_pjsip_registrar.c:432 register_aor_core: Unable to bind contact 'sip:6162480909-300@192.168.201.105:59758' to AOR '6162480909-300' Registration has failed at this point. I can offer the following: A dump of the database schema that alembic is creating. extconfig.config sorcery.conf Thanks Bryant From: "Joshua Colp" Sent: Friday, September 15, 2017 9:56 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime pjsip issues On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote: > Joshua > > That is the interesting part of it. We took our configs and database > tables from our working 13.12.2 deployments and tried to use them with > our > new 13.17.1 deployments and we are having issues where the tables are not > working. On the new server asterisk keeps saying it can't find the AORS > entries when we purge the sorcery memory cache it starts finding the aors > but then it says it cant find the auths. > > The wired thing is when it says it can't find the aors and auths entries > it does not show it is looking for the values in the aors and auth fields > from the endpoints tables. It keeps putting the value from the endpoints > id > field as the entries it can't find. >
Re: [asterisk-users] Realtime pjsip issues
Joshua We have completed more testing this morning and when we remove the realtime cache options from the sorcery file the endpoints complete registration, but we pjsip show/list does not offer any feed back at all, We also can't send any pjsip send notify commands as they say they don't have an endpoint there. Something has changed in the cache part of the system that is breaking the system in some manner for us with the current version and we are out of ideas. Thanks Bryant Joshua That is the interesting part of it. We took our configs and database tables from our working 13.12.2 deployments and tried to use them with our new 13.17.1 deployments and we are having issues where the tables are not working. On the new server asterisk keeps saying it can't find the AORS entries when we purge the sorcery memory cache it starts finding the aors but then it says it cant find the auths. The wired thing is when it says it can't find the aors and auths entries it does not show it is looking for the values in the aors and auth fields from the endpoints tables. It keeps putting the value from the endpoints id field as the entries it can't find. One point of note the tables we used and created for pjsip back when we setup the 13.12.2 version are not what is currently being created when we run alembic now.. Also the contact table from alembic creation process does not work we get insert errors inside of asterisk when contact entry attempts are being crated. It shows a number of fields that are not there in the created tables. This is the foundation of my issues. I really have to resolve them in some manner so I can mover forward with getting these new systems into production. Any assistance is appreciated. Thanks Bryant From: "Joshua Colp" Sent: Thursday, September 14, 2017 4:34 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime pjsip issues On Thu, Sep 14, 2017, at 05:27 PM, Bryant Zimmerman wrote: > This appears to be some kind of cache issue. > We have been doing caching with earlier versions of asterisk 13 on the > pjsip realtime, but now for some reason > The items only show up the first time we use pjsip list/show and then > they > are wiped. I see a new full cache option and that appears to make a > difference, but it is unclear what is going on. In effect it appears that > items loaded from a database for pjsip must be fully cached or you can't > look up any data. > > Why has a change of this magnitude been put into an LTS? > What is the best practices. I see in some of the wikis cache > suggestions. > What are others really seeing? There haven't been any changes made except for bug fixes to the sorcery memory cache, certainly no behavior changes. In fact the implementation is the same between 13 and 14 except for a single line addition. What is your sorcery.conf for both? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime pjsip issues
Joshua That is the interesting part of it. We took our configs and database tables from our working 13.12.2 deployments and tried to use them with our new 13.17.1 deployments and we are having issues where the tables are not working. On the new server asterisk keeps saying it can't find the AORS entries when we purge the sorcery memory cache it starts finding the aors but then it says it cant find the auths. The wired thing is when it says it can't find the aors and auths entries it does not show it is looking for the values in the aors and auth fields from the endpoints tables. It keeps putting the value from the endpoints id field as the entries it can't find. One point of note the tables we used and created for pjsip back when we setup the 13.12.2 version are not what is currently being created when we run alembic now.. Also the contact table from alembic creation process does not work we get insert errors inside of asterisk when contact entry attempts are being crated. It shows a number of fields that are not there in the created tables. This is the foundation of my issues. I really have to resolve them in some manner so I can mover forward with getting these new systems into production. Any assistance is appreciated. Thanks Bryant From: "Joshua Colp" Sent: Thursday, September 14, 2017 4:34 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime pjsip issues On Thu, Sep 14, 2017, at 05:27 PM, Bryant Zimmerman wrote: > This appears to be some kind of cache issue. > We have been doing caching with earlier versions of asterisk 13 on the > pjsip realtime, but now for some reason > The items only show up the first time we use pjsip list/show and then > they > are wiped. I see a new full cache option and that appears to make a > difference, but it is unclear what is going on. In effect it appears that > items loaded from a database for pjsip must be fully cached or you can't > look up any data. > > Why has a change of this magnitude been put into an LTS? > What is the best practices. I see in some of the wikis cache > suggestions. > What are others really seeing? There haven't been any changes made except for bug fixes to the sorcery memory cache, certainly no behavior changes. In fact the implementation is the same between 13 and 14 except for a single line addition. What is your sorcery.conf for both? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime pjsip issues
This appears to be some kind of cache issue. We have been doing caching with earlier versions of asterisk 13 on the pjsip realtime, but now for some reason The items only show up the first time we use pjsip list/show and then they are wiped. I see a new full cache option and that appears to make a difference, but it is unclear what is going on. In effect it appears that items loaded from a database for pjsip must be fully cached or you can't look up any data. Why has a change of this magnitude been put into an LTS? What is the best practices. I see in some of the wikis cache suggestions. What are others really seeing? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Bryant Zimmerman" Sent: Thursday, September 14, 2017 2:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Realtime pjsip issues We are having an issue where on the latest version of asterisk when configuration pjsip via realtime. we do a pjsip list endpoints it shows our endpoints but lists them as invalid. When we do the pjsip list endpoints again it shows no objects. This applies to pjsip list aors as well. We did not have this issue on our older asterisk 13 installs. My guess is something has changed with pjsip and realtime. Anyone have any ideas where I can start. We have tried a number of things already and would love some suggestions. Thanks zktech -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime pjsip issues
We are having an issue where on the latest version of asterisk when configuration pjsip via realtime. we do a pjsip list endpoints it shows our endpoints but lists them as invalid. When we do the pjsip list endpoints again it shows no objects. This applies to pjsip list aors as well. We did not have this issue on our older asterisk 13 installs. My guess is something has changed with pjsip and realtime. Anyone have any ideas where I can start. We have tried a number of things already and would love some suggestions. Thanks zktech -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] softphone instead of desktop phones
Thomas Bria is by counterpath Bryant From: "Matt Riddell (lists)" Sent: Saturday, April 29, 2017 11:50 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] softphone instead of desktop phones I use Bria on all of the above. Kind regards, Matt On Apr 29, 2017, at 10:35 AM, Thomas wrote: Hello, Iam lookong for an Softphone for iPhor oder Android smartphone using togehter with an headset. I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP phone. Is there an better softphone? Or are there softphone solutions for PC desktop MAC or Android with an headset? I want to save cost for desktop phones. thanks Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] softphone instead of desktop phones
Thomas I have found you will likely end up spending more with softphones. You have to purchase a good headset and quality does not come inexpensive with headsets. We find we will pay $80 to $100 dollars plus for a fair quality headset. You also have to support deployment to devices. Which means in many cases you must touch the device for maintenance. The best softphone I have found is by counter path. They have versions for PC, MAC, Android and IOS. You will spend about $60.00 / device for the software. They have a fee edition for some devices, but not as feature rich. I have found it much more successful to deploy something like a Grand Stream Phone $100.00 for a nice feature phone, and I get convenient remote provisioning, Users never have username and passwords so security is ensured, Quality is good, I don't have to worry about phone issues if their pc or smartphone is acting up for the most part they just work. There are phones for as low as $50.00 from grand stream with high quality audio just fewer features. Softphones can work well, but when looking at trouble tickets over the last 13 years. I get more from my softphone users then from my desk phone users. And I can hear the quality difference when I talk to the softphone users it is such a wild card. How important is consistent quality to you? Users change headsets, move mic positions and the caller on the other end gets the short end of it. Also remember virus scans, patches, updates, system reboots, wifi signal strength, cell signal (cell ip calls) cheep Bluetooth headsets and the list goes on and on. The counterpath software is as good as I have found, but there are additional variables you have to look at. Which is the best for your use case and the cheapest? You have to figure out your formula make sure to look at all the costs and factors how much is support time worth? In very controlled environments where you can have consistent control and good quality headsets such as call centers soft clients can work well, but that is not how you started out you are looking for quality on the cheep. Desk phones are cheep and in most cases just work and offer consistent quality. If others have found different I look forward to seeing their responses. This is a great question thanks for asking it Thomas. Best of luck Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Thomas" Sent: Saturday, April 29, 2017 11:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] softphone instead of desktop phones Hello, Iam lookong for an Softphone for iPhor oder Android smartphone using togehter with an headset. I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP phone. Is there an better softphone? Or are there softphone solutions for PC desktop MAC or Android with an headset? I want to save cost for desktop phones. thanks Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED]
In most instances the company being called is not charging the caller for their phone serves. That is the callers service provider, and once the answer is issued the call is up. This only makes senses if the company being called is providing services and charging a per min rate for that service. They would not charge the customer for the hold time waiting for a rep to come on the line. This could all be done by creating billing records from cel logs. These can log events such as channel start and answer by an extension, transfers and hangups. As Samy Go stated a good way to reduce charges to the caller would be to offer call back options. So when a rep is available the system would call the original caller back. Telecom networks around the world are just not designed to offer delayed billing. Legislating that requirement would require world wide overhauls of the networks as well as treaties. In some areas you also have to pay ring time. That is a novel idea to actually pay for a resource you are using when you use it. That is a little too capitalistic for some. Bryant From: "SamyGo" Sent: Wednesday, March 29, 2017 9:52 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED] Hi, Just trying to figure out how is this solved ? by involving multiple telcos in the loop and asking them to not charge based on 200 OK/Answer!? As far as I know people have designed Queue/CallCenter platforms who upon entering a number in queue just state them their number in queue and approx time before they'll be contacted and drop the call. This all can be done within Progress. As soon as their turn comes the CallCenter platform automatically triggers the call to them and get them connected with an agent. This is the way I can understand as nobody waiting in the queue but people in the waiting-list. Since Queue has to "answer" the call first before doing anything once the signal to Answer is triggered technically that marks the start of billing for everyone. Regards, Sammy On Wed, Mar 29, 2017 at 7:25 AM, Olivier wrote: Thank you very much, Max, for this valuable and informative answer. Offline billing must be quite complex to set up as several telco may be involved (or origination,transit or termination). Moving to normal landline fare seems much simpler ! Thanks again2017-03-28 21:41 GMT+02:00 Max Grobecker : Hi, in Germany, this kind of regulation is in effect for phone numbers which cost more than a normal landline call. The regulation states, that the waiting time must not be charged to the customer. Most companies implemented this by simply switching their telephone numbers to those, which are charged per call (so there's no difference in price between waiting for someone to pick up or being connected to someone) ;-) Or they decided to use a normal landline phone number for which this regulation does not apply. The second method was to not answer the call before really connected to a person on the queue and using Early Media as you mentioned. But: The maximum length of this Early Media stream is in most telephone networks limited to somewhat around 90 to 180 seconds, then the call gets disconnected by the network. I'm not very familiar with regulations and numbering plans in France, but maybe there's also something called "offline billing". Using this, your call is not billed by the caller's telephone company until you send them the amount of time that should be billed for a specific call. Your best choice will be, that - if you ever get those regulations - you should rely on what your telephone number provider tells you to do ;-) Greetings Max Am 28.03.2017 um 15:24 schrieb Olivier: > Hello, > > In France, years ago, there was some discussions about a new regulation forcing some providers to not charge anything to callers while those are waiting for a call center agent to become available. > Once caller and agent are on call with each other, nominal charging applies. > > No matter if those discussions ever did or didn't change current regulation, I wonder which dialplan statements could technically comply this dual billing requirement ? > > > same = n,Progress() > same = n,Queue(whatever,...,macro-option, ...) > > To me, coupling Progress app with Queue's macro or gosub option like above, would let a sysadmin answer a queued call. > Doing so, time spent before connection with queue agent should not be billed to anyone (caller nor callee), while time spent after connection is billed normaly. > > 1. Should this work ? Am I missing something ? > > 2. Is there an alternative way to implement this ? > > 3. Comments ? Suggestions ? > > Regards > > -- _ -- Bandwidth and Colocation Provided
Re: [asterisk-users] WebRTC - Transport Issues. - Solved
Josh Thank you for the confirmation on this. The captures do confirm that I am using the wss. What was throwing me was I have only udp and wss in the transports and then the Primary once connected was showing the ws. At first I thought I was doing something wrong and the traffic was flowing unencrypted. You confirmed what I had hoped that the wss was just showing the underlying ws transport. A big thanks. We are excited to finally getting our webrtc test application out to some customers. Have a great week. Bryant From: "Joshua Colp" Sent: Sunday, March 12, 2017 7:35 PM On Sat, Mar 11, 2017, at 09:52 PM, Bryant Zimmerman wrote: > Hey all. I have webrtc up and running with asterisk 11. All is going well > with TLS now working. > At least I hope it is using TLS and wss. Based on what I am seeing I > have > UDP, WSS listed in the Allowed transports, but every time I connect the > Primary transport shows WS.. Why is this? Am I actually running ws in > wss > mode? You are using WSS (the Contact line has transport=wss which indicates it). Both WS and WSS will show "WS" for the Primary Transport. Another way to tell is to look at the SIP traffic and check the Via header for WSS. You can also check a packet capture. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WebRTC - Transport Issues.
Hey all. I have webrtc up and running with asterisk 11. All is going well with TLS now working. At least I hope it is using TLS and wss. Based on what I am seeing I have UDP, WSS listed in the Allowed transports, but every time I connect the Primary transport shows WS.. Why is this? Am I actually running ws in wss mode? Prim.Transp. : WS Allowed.Trsp : UDP,WSS Def. Username: 6167761066.2011 SIP Options : (none) Codecs : (ulaw) Codec Order : (ulaw:20) Auto-Framing : No Status : OK (71 ms) Useragent: SIP.js/0.7.7 Reg. Contact : sip:fed97qgu@192.0.2.35;transport=wss Any Insights would be appreciated. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code. - Solved
I figured this out. I had to set the outofcall_message_context = messages on the actual peer. It was not good enough to set in the sip.conf Thanks Bryant From: "Bryant Zimmerman" Sent: Friday, March 10, 2017 11:39 AM Jean Thank you for your response. I have the options you suggested already set, and I am still not getting the dialplan to trigger. The message is being sent but nothing. I have tried with the auth both set to no and yes as well. accept_outofcall_message = yes outofcall_message_context = messages auth_message_requests = yes Thanks Bryant From: "Jean Aunis" Sent: Friday, March 10, 2017 2:24 AM This is not a SIP NOTIFY but a SIP MESSAGE (the first line in the logs is not related to the few next ones). If you are using chan_sip, you have to activate out of call messages in sip.conf : accept_outofcall_message=yes outofcall_message_context=messages Then in extensions.conf, define a context "messages" with the appropriate extensions (to stick to your example, it will be 16162995607) and use the function MESSAGE to retrieve the SMS content. Best regards Jean Aunis Le 10/03/2017 à 00:21, Bryant Zimmerman a écrit : I am trying to send SMS from my grandstream GXV3240 Asterisk receives the message in a NOTIFY block. How can I get asterisk to run dialplan code when receiving these Notify SMS Message Blocks. I can then route them to my SMS provider. Any ideas are appreciated. Below is debug of a message sent from the phone when received no dialplan code is triggered. I am wounding if I need to modify some setting in sip.conf or the peer config. Incomming SMS from my vendor works without issue and is transmitted to the phone. <-> --- (10 headers 0 lines) --- Really destroying SIP dialog '3927411c7fe967886df6c8d0410d4...@xxx.xxx.xxx.xxx:5060' Method: NOTIFY <--- SIP read from UDP:XXX.XXX.XXX.XXX:57568 ---> MESSAGE sip:16162995...@vgw0005.granddial.net SIP/2.0 Via: SIP/2.0/UDP 192.168.201.104:20093;branch=z9hG4bK1738682353;rport From: ;tag=1683585926 To: Call-ID: 1662412698-20093-...@bjc.bgi.cab.bae CSeq: 9430 MESSAGE Contact: Max-Forwards: 70 User-Agent: Grandstream GXV3240 1.0.3.158 Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: text/plain; charset=UTF-8 Content-Length: 5 Test Message SMS <-> Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code.
Jean Thank you for your response. I have the options you suggested already set, and I am still not getting the dialplan to trigger. The message is being sent but nothing. I have tried with the auth both set to no and yes as well. accept_outofcall_message = yes outofcall_message_context = messages auth_message_requests = yes Thanks Bryant From: "Jean Aunis" Sent: Friday, March 10, 2017 2:24 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code. This is not a SIP NOTIFY but a SIP MESSAGE (the first line in the logs is not related to the few next ones). If you are using chan_sip, you have to activate out of call messages in sip.conf : accept_outofcall_message=yes outofcall_message_context=messages Then in extensions.conf, define a context "messages" with the appropriate extensions (to stick to your example, it will be 16162995607) and use the function MESSAGE to retrieve the SMS content. Best regards Jean AunisLe 10/03/2017 à 00:21, Bryant Zimmerman a écrit : I am trying to send SMS from my grandstream GXV3240 Asterisk receives the message in a NOTIFY block. How can I get asterisk to run dialplan code when receiving these Notify SMS Message Blocks. I can then route them to my SMS provider. Any ideas are appreciated. Below is debug of a message sent from the phone when received no dialplan code is triggered. I am wounding if I need to modify some setting in sip.conf or the peer config. Incomming SMS from my vendor works without issue and is transmitted to the phone. <-> --- (10 headers 0 lines) --- Really destroying SIP dialog '3927411c7fe967886df6c8d0410d4...@xxx.xxx.xxx.xxx:5060' Method: NOTIFY <--- SIP read from UDP:XXX.XXX.XXX.XXX:57568 ---> MESSAGE sip:16162995...@vgw0005.granddial.net SIP/2.0 Via: SIP/2.0/UDP 192.168.201.104:20093;branch=z9hG4bK1738682353;rport From: ;tag=1683585926 To: Call-ID: 1662412698-20093-...@bjc.bgi.cab.bae CSeq: 9430 MESSAGE Contact: Max-Forwards: 70 User-Agent: Grandstream GXV3240 1.0.3.158 Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: text/plain; charset=UTF-8 Content-Length: 5 Test Message SMS <-> Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code.
I am trying to send SMS from my grandstream GXV3240 Asterisk receives the message in a NOTIFY block. How can I get asterisk to run dialplan code when receiving these Notify SMS Message Blocks. I can then route them to my SMS provider. Any ideas are appreciated. Below is debug of a message sent from the phone when received no dialplan code is triggered. I am wounding if I need to modify some setting in sip.conf or the peer config. Incomming SMS from my vendor works without issue and is transmitted to the phone. <-> --- (10 headers 0 lines) --- Really destroying SIP dialog '3927411c7fe967886df6c8d0410d4...@xxx.xxx.xxx.xxx:5060' Method: NOTIFY <--- SIP read from UDP:XXX.XXX.XXX.XXX:57568 ---> MESSAGE sip:16162995...@vgw0005.granddial.net SIP/2.0 Via: SIP/2.0/UDP 192.168.201.104:20093;branch=z9hG4bK1738682353;rport From: ;tag=1683585926 To: Call-ID: 1662412698-20093-...@bjc.bgi.cab.bae CSeq: 9430 MESSAGE Contact: Max-Forwards: 70 User-Agent: Grandstream GXV3240 1.0.3.158 Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: text/plain; charset=UTF-8 Content-Length: 5 Test Message SMS <-> Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail2ban Asterisk 13.13.1
John V Are you using pjsip? We are have several test servers and I just checked my /etc/fail2ban/filter.d/asterisk.conf and it is not updated for pjsip implementations. Looking at the security log files and the regex I noticed that some items are being banned but others are not due to changes in the messages for pjsip. Anyone got an updated asterisk.conf for fail2ban. Bryant From: "Telium Technical Support" Sent: Wednesday, March 1, 2017 9:54 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] fail2ban Asterisk 13.13.1 If this is a small site, I recommend you download the free version of SecAst (www.telium.ca) and replace fail2ban. SecAst does NOT use the log file, or regexes, to match etc.instead it talks to Asterisk through the AMI to extract security information. Messing with regexes is a losing battle, and the lag in reading logs can allow an attacker 100+ registration attempts before fail2ban even does anything (assuming the IP is exposed in the Asterisk log). If this is a large install then post in the commercial list for more information. -Raj- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support Sent: Wednesday, March 1, 2017 2:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] fail2ban Asterisk 13.13.1 It's possible that you need to increase the value of 'findtime' to something greater than 300 secs. You also may want to set "timestamp = yes" in asterisk.conf so each line in the CLI will be time stamped. Time stamping it will be the definitive determination on whether or not the 'findtime' is the culprit. Regards; John V. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motty Cruz Sent: Wednesday, March 01, 2017 01:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] fail2ban Asterisk 13.13.1 Hello, fail2ban does not ban offending IP. NOTICE[29784] chan_sip.c: Registration from '"user3"' failed for 'offending-IP:53417' - Wrong password NOTICE[29784] chan_sip.c: Registration from '"user3"' failed for 'offending-IP:53911' - Wrong password # A host is banned if it has generated "maxretry" during the last "findtime" # seconds. findtime = 300 [asterisk-iptables] enable = true port = 5060,5061 filter = asterisk action = iptables-allports[name=ASTERISK, protocol=all] sendmail[name=ASTERISK, dest=mo...@email.com, sender=fail2...@asterisk-ip.com] #action = %(banaction)s[name=%(__name__)s-tcp, port="%(port)s", protocol="tcp", chain="%(chain)s", actname=%(banaction)s-tcp] %(banaction)s[name=%(__name__)s-udp, port="%(port)s", protocol="udp", chain="%(chain)s", actname=%(banaction)s-udp] %(mta)s-whois[name=%(__name__)s, dest="%(destemail)s"] logpath = /var/log/asterisk/messages maxretry = 3 findtime = 300 bantime = -1 in filter.d asterisk.conf failregex = ^%(__prefix_line)s%(log_prefix)s Registration from '[^']*' failed for '(:\d+)?' - (Wrong password|Username/auth name mismatch|No matching peer found|Not a local domain|Device does not match ACL|Peer is not supposed to register|ACL error \(permit/deny\)|Not a local domain)$ ^%(__prefix_line)s%(log_prefix)s Call from '[^']*' \(:\d+\) to extension '[^']*' rejected because extension not found in context ^%(__prefix_line)s%(log_prefix)s Host failed to authenticate as '[^']*'$ ^%(__prefix_line)s%(log_prefix)s No registration for peer '[^']*' \(from \)$ ^%(__prefix_line)s%(log_prefix)s Host failed MD5 authentication for '[^']*' \([^)]+\)$ ^%(__prefix_line)s%(log_prefix)s Failed to authenticate (user|device) [^@]+@\S*$ ^%(__prefix_line)s%(log_prefix)s hacking attempt detected ''$ ^%(__prefix_line)s%(log_prefix)s SecurityEvent="(FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPassword)",EventTV="([\d-]+|%(iso8601)s)",Severity="[\w]+",Service="[\w]+",EventVersion="\d+",AccountID="(\d*|)",SessionID=".+",LocalAddress="IPV[46]/(UDP|TCP|WS)/[\da-fA-F:.]+/\d+",RemoteAddress="IPV[46]/(UDP|TCP|WS)//\d+"(,Challenge="[\w/]+")?(,ReceivedChallenge="\w+")?(,Response="\w+",ExpectedResponse="\w*")?(,ReceivedHash="[\da-f]+")?(,ACLName="\w+")?$ ^%(__prefix_line)s%(log_prefix)s "Rejecting unknown SIP connection from "$ ^%(__prefix_line)s%(log_prefix)s Request (?:'[^']*' )?from '[^']*' failed for '(?::\d+)?'\s\(callid: [^\)]*\) - (?:No matching endpoint found|Not match Endpoint(?: Contact)? ACL|(?:Failed|Error) to authenticate)\s*$ failregex = NOTICE.* .*: Registration from '.*' failed for '' - Wrong password NOTICE.* .*: Registration from '.*' failed for ':.*' - No matching peer found
Re: [asterisk-users] pjsip realtime - endpoints not loading - Solved
It appears that res_odbc.so does not always load fast enough to allow the realtime mappings in the extconfig.conf to complete successfully at startup thus stopping the first load of the pjsip endpoints and other pjsip values. The resolution for this is to preload the res_odbc.so and res_config_odbc.so in the modules.conf. The realtime mappings then appear to complete correctly during startup and allows all the pjsip data to load correctly. Thanks Bryant Sent: Wednesday, December 21, 2016 9:12 AM Subject: [asterisk-users] pjsip realtime - endpoints not loading. We are continuing to test our asterisk 13 pjsip deployments. I am running into an issue that I am assuming is a configuration problem, and am hoping someone can point me in the right direction. We are running pjsip in real-time mode using a database to store all the endpoint records. Our endpoint records with our carrier do not support registration. The issue I am having is when asterisk starts none of the non registration endpoints become available. They will not allow calls inbound or acknowledge qualify's. To get them to come on line we have to do a pjsip show endpoints, and then all works until asterisk is restarted. Is there any way to get the endpoints to load without manually doing the pjsip show endpoints? Any input is appreciated. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pjsip realtime - endpoints not loading.
We are continuing to test our asterisk 13 pjsip deployments. I am running into an issue that I am assuming is a configuration problem, and am hoping someone can point me in the right direction. We are running pjsip in real-time mode using a database to store all the endpoint records. Our endpoint records with our carrier do not support registration. The issue I am having is when asterisk starts none of the non registration endpoints become available. They will not allow calls inbound or acknowledge qualify's. To get them to come on line we have to do a pjsip show endpoints, and then all works until asterisk is restarted. Is there any way to get the endpoints to load without manually doing the pjsip show endpoints? Any input is appreciated. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax faling on PJSip
I am working on moving from version 11 to version 13 for my fax applications. We are bumping into an issue where the bulk of the T38 faxes are failing. The sending test switch is reporting COMREC_ERR_TRANSMIT_PHASE These same faxes succeed on the 11 version of asterisk. I am wondering if there are any ideas? COMREC_ERR_TRANSMIT_PHASE Both servers are running the same version of spandsp. The dialplan code is the same on both. The only difference is the versions of asterisk and pjsip on the 13 platform. Any ideas would be appreciated. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 T.38 Version 3?
Does anyone know if Asterisk 13 will support T.38 Version 3? ? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip transports from database.
On Friday, November 4, 2016 10:20 AM - Joshua Colp wrote: >>On Fri, Nov 4, 2016, at 10:26 AM, Bryant Zimmerman wrote: >> Hey all >> >> I am trying to configure all my pjsip transports form a database table. >> The issue I am running into is that pjsip is auto binding to 0.0.0.0:5060 >> before it reads my list of transports from the database. This means that >> my >> entries for port 5060 are already bound and the settings in the database >> are not loaded. >> >> When loading the transport form the .conf file it works as expected and >> does not do an auto binding, but uses what is in the .conf >> >> Is there a way to have asterisk pjsip hold the default binding override >> until after it has checked the database when sourcery .conf configures a >> transport location other then pjsip.conf? >>PJSIP has no auto binding or default binding. It will only bind to what >>Is configured. Do you have it in both .conf and in realtime? Do you also >>have chan_sip loaded? Joshua You were correct. There was an old chan_sip.so in the bin folder that was being auto loaded. It was binding to 0.0.0.0:5060 causing the transports from the database for pjsip to fail. I forced down asterisk and deleted the chan_sip.so from the bin folder and the issue resolved. Looks like I need to go through and clean up old garbage from an earlier build so I don't get caught in the future. I also added a noload for chan_sip.so just incase one ever gets dropped back in the folder. Much thanks for the direction here I spent a lot of time trying to figure out where the binding was coming from. Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pjsip transports from database.
Hey all I am trying to configure all my pjsip transports form a database table. The issue I am running into is that pjsip is auto binding to 0.0.0.0:5060 before it reads my list of transports from the database. This means that my entries for port 5060 are already bound and the settings in the database are not loaded. When loading the transport form the .conf file it works as expected and does not do an auto binding, but uses what is in the .conf Is there a way to have asterisk pjsip hold the default binding override until after it has checked the database when sourcery .conf configures a transport location other then pjsip.conf? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP - State of the art
I agree the multi-domain environment is a nice idea, but too many endpoints don't properly support. We to use a prefix in the SIP username for multi-domain environments. Thanks Bryant From: "Ludovic Gasc" Sent: Sunday, July 17, 2016 5:20 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] PJSIP - State of the art 2016-07-17 14:30 GMT+02:00 Annus Fictus : The main idea of the new channel was working on a multi-domain environment For now, to my experience, it's more future-proof compliant to use a prefix in the SIP username than multi-domain environment. Even if the multi-domain support was perfect in Asterisk, we tested some crappy SIP endpoints where in fact, even if you configure a domain name everywhere in the configuration, you have only IPs in SIP packets. We have that on production for our cloud plateform, it works pretty well and also simplify whitelabel handling. Moreover, if you have a good provisioning support, it will be invisible for your users. When I see the time needed to really use on production the SNI feature in SSL, and you have only 5 majors HTTP endpoints (aka Web browsers). In the SIP world, I'm not sure you can use multi domain except if you can force the SIP endpoints used by your clients. , have more then one device registered with same credentials and have more stability. Since 13.9.1, we have a better experience of pjsip. Nevertheless, not yet massively used on production for now, we planned to migrate endpoint by endpoint to minimize the risk. Be Better still with Asterisk 1.11.X? Maybe you could use Asterisk 13 with chan_sip to start, it works pretty well and already think to support chan_pjsip in the same time. The benefit to think about that if one day you need to use an alternative channel like chan_iax2, it should be easier to implement for you. Regards -- _ -- 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] Call File - CPU spikes
I am working on a project that we are seeing a 100% CPU spike when we move 50 calls files to the folder. We are running pjsip and asterisk 13..It holds the spike for several minutes Are there any tunable that may help with this? Thanks Bryant -- _ -- 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] ? Re: Recommendations for free virtual server tech and Asterisk? (Ikka Tirtawidjaja)
Hyper-V works well we run both OpenSuse and Debian with asterisk on it is rock solid, and it is free if you use the Hyper-V Server Version. Bryant From: "Saint Michael" Sent: Saturday, April 9, 2016 1:23 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] ? Re: Recommendations for free virtual server tech and Asterisk? (Ikka Tirtawidjaja) ?OpenVZ is useless for Asterisk or any other resource intensive application. OpenVZ was built from a hosting provider point if view, and if you exceed any of the counters, dozens of them, they system will kill your app immediately. It is almost impossible to build a VPS that will use all the resources of the machine. The only container technology that works for Asterisk is LXC, better implemented by Ubuntu on the server side. Centos 7 is behind in the LXC version and it is part of the core OS, but found in a repository. Yo need kernel 4.X to make it works flawlessly. ? -- _ -- 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 - Call Bridge issue.
Even when using the U option just issuing the Answer does not seem to
always work. I end up having to play a prompt of some sort to force the
answer.. There has to be some kind of bug going on here.
Thanks
Bryant
From: "Bryant Zimmerman"
Sent: Thursday, March 31, 2016 6:54 PM
To: brya...@zktech.com, "Asterisk Users Mailing List - Non-Commercial
Discussion"
Subject: re: [asterisk-users] Asterisk 13 - Call Bridge issue.
----
From: "Bryant Zimmerman"
Sent: Thursday, March 31, 2016 6:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk 13 - Call Bridge issue.
I have the following scenario.
Call file calls 1st party.
When connected give called party option to connect to second party.
Issue Dial to second party. Caller answers and the two are bridged
together.
My issue is that 4 out of 5 calls fail to bridge the audio.
Am I missing something or is there some kind of bug? Here is my test
dialplan
;Dialer Base Code Files.
;Variables are sent in from .call file
[calluser-intake]
exten => s,1,NoOp(Start Call Intake)
exten => s,2,NoOp(Setup any vars)
exten => s,n,Set(_g_pmtPath=/vapp/dialerprompts/)
exten => s,n,NoOp(What is Path = ${g_pmtPath})
exten => s,n,NoOp(Read Call File Vars)
exten => s,n,NoOp(Dial To - ${l_DialTo})
exten => s,n,NoOp(Proxy - Proxy.${l_Proxy})
exten => s,n,NoOp(Carrier Trunk - ${l_Carrier})
exten => s,n,Set(_l_CallerIDnum=${CALLERID(num)})
exten => s,n,Set(CALLERID(num)=${g_SIPUser})
exten =>
s,n,Dial(PJSIP/${l_DialTo}@proxy_${l_Proxy},30,b(dialer-header^s^1)G(dialer-
playmsg^s^1))
[dialer-header]
exten => s,1,Set(PJSIP_HEADER(add,X-Carrier)=${l_Carrier})
same => n,Set(PJSIP_HEADER(add,X-CallerID)=${l_CallerIDnum})
same => n,NoOp(X-Carrier = ${PJSIP_HEADER(read,X-Carrier)})
same => n,Set(CONNECTEDLINE(number,i)=vap_002)
same => n,DumpChan(1)
same => n,Return()
[dialer-playmsg]
exten => s,1,Goto(hold,1)
same => n,NoOp(Enter Play Message)
same => n,NoOp(Path = ${g_pmtPath})
same => n,SayAlpha(${g_SIPUser})
same => n,BackGround(${g_pmtPath}Intro)
same => n,WaitExten(60)
exten => 2,1,NoOp(Dial Through)
same => n,Set(_l_CallerIDnum=616831)
same => n,Set(_l_Carrier=0001)
same => n,Set(l_DialTo=6167761066)
same => n,Set(l_Proxy=002)
same => n,Dial(PJSIP/${l_DialTo}@proxy_001,30,b(dialer-header^s^1))
exten => _X,1,NoOp(Digit Entry)
exten => _X,n,NoOp(Log Response)
exten => _X,n,Playback(${g_pmtPath}YouPressed)
exten => _X,n,SayNumber(${EXTEN})
exten => hold,1,NoOp(Park Called)
exten => hold,n,While($[1 < 5])
exten => hold,n,Wait(90)
exten => hold,n,EndWhile
Any ideas on why the media would not flowing after it sates they bridge
has completed
Another point. If I use a b option in the second dial. to call another
context on connect of the second call. I get audio played on that both
caller and callee channels.
Thanks
Bryant
Ok it appears that the channel is not answering when it bridges the two
calls together.
If I use the U option to gosub to a context to force an Answer() before
the bridge then things seem to work. I also tried the lower case "a" option
to force the answer and nothing happens with it appears to be ignored. ..
So the U option with a gosub to an Answer seems to be the only way to get
this to work...
This seems like a bug. Should the called channel answer when a call is
made with the Dial() function? Can anyone chime in on this one.
Note: Current systems are on Asterisk 13.5.0 (So if this was a bug has it
been fixed in the latest release.) I did not see anything in the change
logs that I would attribute to this.
Thanks
Bryant
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Re: [asterisk-users] Asterisk 13 - Call Bridge issue.
From: "Bryant Zimmerman"
Sent: Thursday, March 31, 2016 6:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk 13 - Call Bridge issue.
I have the following scenario.
Call file calls 1st party.
When connected give called party option to connect to second party.
Issue Dial to second party. Caller answers and the two are bridged
together.
My issue is that 4 out of 5 calls fail to bridge the audio.
Am I missing something or is there some kind of bug? Here is my test
dialplan
;Dialer Base Code Files.
;Variables are sent in from .call file
[calluser-intake]
exten => s,1,NoOp(Start Call Intake)
exten => s,2,NoOp(Setup any vars)
exten => s,n,Set(_g_pmtPath=/vapp/dialerprompts/)
exten => s,n,NoOp(What is Path = ${g_pmtPath})
exten => s,n,NoOp(Read Call File Vars)
exten => s,n,NoOp(Dial To - ${l_DialTo})
exten => s,n,NoOp(Proxy - Proxy.${l_Proxy})
exten => s,n,NoOp(Carrier Trunk - ${l_Carrier})
exten => s,n,Set(_l_CallerIDnum=${CALLERID(num)})
exten => s,n,Set(CALLERID(num)=${g_SIPUser})
exten =>
s,n,Dial(PJSIP/${l_DialTo}@proxy_${l_Proxy},30,b(dialer-header^s^1)G(dialer-
playmsg^s^1))
[dialer-header]
exten => s,1,Set(PJSIP_HEADER(add,X-Carrier)=${l_Carrier})
same => n,Set(PJSIP_HEADER(add,X-CallerID)=${l_CallerIDnum})
same => n,NoOp(X-Carrier = ${PJSIP_HEADER(read,X-Carrier)})
same => n,Set(CONNECTEDLINE(number,i)=vap_002)
same => n,DumpChan(1)
same => n,Return()
[dialer-playmsg]
exten => s,1,Goto(hold,1)
same => n,NoOp(Enter Play Message)
same => n,NoOp(Path = ${g_pmtPath})
same => n,SayAlpha(${g_SIPUser})
same => n,BackGround(${g_pmtPath}Intro)
same => n,WaitExten(60)
exten => 2,1,NoOp(Dial Through)
same => n,Set(_l_CallerIDnum=616831)
same => n,Set(_l_Carrier=0001)
same => n,Set(l_DialTo=6167761066)
same => n,Set(l_Proxy=002)
same => n,Dial(PJSIP/${l_DialTo}@proxy_001,30,b(dialer-header^s^1))
exten => _X,1,NoOp(Digit Entry)
exten => _X,n,NoOp(Log Response)
exten => _X,n,Playback(${g_pmtPath}YouPressed)
exten => _X,n,SayNumber(${EXTEN})
exten => hold,1,NoOp(Park Called)
exten => hold,n,While($[1 < 5])
exten => hold,n,Wait(90)
exten => hold,n,EndWhile
Any ideas on why the media would not flowing after it sates they bridge
has completed
Another point. If I use a b option in the second dial. to call another
context on connect of the second call. I get audio played on that both
caller and callee channels.
Thanks
Bryant
Ok it appears that the channel is not answering when it bridges the two
calls together.
If I use the U option to gosub to a context to force an Answer() before
the bridge then things seem to work. I also tried the lower case "a" option
to force the answer and nothing happens with it appears to be ignored. ..
So the U option with a gosub to an Answer seems to be the only way to get
this to work...
This seems like a bug. Should the called channel answer when a call is
made with the Dial() function? Can anyone chime in on this one.
Note: Current systems are on Asterisk 13.5.0 (So if this was a bug has it
been fixed in the latest release.) I did not see anything in the change
logs that I would attribute to this.
Thanks
Bryant
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[asterisk-users] Asterisk 13 - Call Bridge issue.
I have the following senerio.
Call file calls 1st party.
When connected give called party option to connect to second party.
Issue Dial to second party. Caller answers and the two are bridged
together.
My issue is that 4 out of 5 calls fail to bridge the audio.
Am I missing something or is there some kind of bug? Here is my test
dialplan
;Dialer Base Code Files.
;Variables are sent in from .call file
[calluser-intake]
exten => s,1,NoOp(Start Call Intake)
exten => s,2,NoOp(Setup any vars)
exten => s,n,Set(_g_pmtPath=/vapp/dialerprompts/)
exten => s,n,NoOp(What is Path = ${g_pmtPath})
exten => s,n,NoOp(Read Call File Vars)
exten => s,n,NoOp(Dial To - ${l_DialTo})
exten => s,n,NoOp(Proxy - Proxy.${l_Proxy})
exten => s,n,NoOp(Carrier Trunk - ${l_Carrier})
exten => s,n,Set(_l_CallerIDnum=${CALLERID(num)})
exten => s,n,Set(CALLERID(num)=${g_SIPUser})
exten =>
s,n,Dial(PJSIP/${l_DialTo}@proxy_${l_Proxy},30,b(dialer-header^s^1)G(dialer-
playmsg^s^1))
[dialer-header]
exten => s,1,Set(PJSIP_HEADER(add,X-Carrier)=${l_Carrier})
same => n,Set(PJSIP_HEADER(add,X-CallerID)=${l_CallerIDnum})
same => n,NoOp(X-Carrier = ${PJSIP_HEADER(read,X-Carrier)})
same => n,Set(CONNECTEDLINE(number,i)=vap_002)
same => n,DumpChan(1)
same => n,Return()
[dialer-playmsg]
exten => s,1,Goto(hold,1)
same => n,NoOp(Enter Play Message)
same => n,NoOp(Path = ${g_pmtPath})
same => n,SayAlpha(${g_SIPUser})
same => n,BackGround(${g_pmtPath}Intro)
same => n,WaitExten(60)
exten => 2,1,NoOp(Dial Through)
same => n,Set(_l_CallerIDnum=616831)
same => n,Set(_l_Carrier=0001)
same => n,Set(l_DialTo=6167761066)
same => n,Set(l_Proxy=002)
same => n,Dial(PJSIP/${l_DialTo}@proxy_001,30,b(dialer-header^s^1))
exten => _X,1,NoOp(Digit Entry)
exten => _X,n,NoOp(Log Response)
exten => _X,n,Playback(${g_pmtPath}YouPressed)
exten => _X,n,SayNumber(${EXTEN})
exten => hold,1,NoOp(Park Called)
exten => hold,n,While($[1 < 5])
exten => hold,n,Wait(90)
exten => hold,n,EndWhile
Any ideas on why the media would not flowing after it sates they bridge
has completed
Another point. If I use a b option in the second dial. to call another
context on connect of the second call. I get audio played on that both
caller and callee channels.
Thanks
Bryant
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Re: [asterisk-users] 2 devices same *actual* extension - can it be done
With Asterisk 13 you may be able to do it with PJSIP using two separate connections on the same AOR I believe you would have two separate endpoints that would register under the same user and auth. If I understand it correctly when you send a call to the AOR both registered endpoints would be rung. I have not tried inbound ring yet, but when I have registered for out bound multiple connections and it seems to work well. Bryant From: "Kevin Long" Sent: Wednesday, March 9, 2016 1:42 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] 2 devices same *actual* extension - can it be done Hello, My company has invested heavily in Counterpath's Stretto provisioning platform for Mobile and Desktop VoIP clients . At this time their system allows 2 devices (for example iPhone + desktop computer) using the same software license per user , which many of our users require. Their provisioning system assumes that both devices will use the same SIP extension for auth however. Normally we would use separate extensions and a follow-me , but if there is any way to use the same extension, I need to figure it out. Thank you, 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] Grandstream Early Dial
Jean
If you moved the exten => _. Lines to the bottom of the context then you
should like be able to get away from having to have two separate contexts. I
use that method quiet often, but was in a hurry to get you a response and did
not think remember that nuance.
I will have to try this as we are a heavy grandstream shop. It has been
something on the list.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: "Jean-Denis Girard"
Sent: Friday, February 19, 2016 11:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Grandstream Early Dial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Bryant,
Thanks for your reply.
It didn't work immediately, I had to create a second context, or else it
was looping between the second and first line. This seems to work:
[earlydial] ; Test Early Dial
exten => _.,1,Set(l_Extension=${EXTEN})
exten => _.,n,Goto(earlydial2,${l_Extension},1)
[earlydial2]
exten => _.,n,Goto(noMatch,1)
exten => noMatch,1, Incomplete(n)
exten => i,1,Goto(noMatch,1)
exten => t,1,Goto(noMatch,1)
exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
same => n,Playback(extension)
same => n,SayDigits(${EXTEN})
same => n,Hangup()
Best regards,
- --
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
Le 19/02/2016 03:31, Bryant Zimmerman a écrit :
> Jean-Denis Girard
>
> I have not used the Incomplete yet, but you might be able to do
> something like this.
>
> [earlydial]
>
> exten => _.,1,Set(l_Extension = ${EXTEN})
> exten => _.,n,Goto(${l_Extension},1)
> exten => _.,n,Goto(noMatch,1)
>
> exten => i,1,Goto(noMatch,1)
>
> exten => noMatch,1, Incomplete(n)
>
> exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
> same => n,Playback(extension)
> same => n,SayDigits(${EXTEN})
> same => n,Hangup()
>
>
> I wrote this in this message and have not tested this so use with
> caution. There may be syntactical issues, but the concept might work f
or
> you.
>
> Bryant
>
> --
- --
> *From*: "Jean-Denis Girard"
> *Sent*: Thursday, February 18, 2016 8:02 PM
> *To*: asterisk-users@lists.digium.com
> *Subject*: Re: [asterisk-users] Grandstream Early Dial
>
> Le 18/02/2016 11:03, Richard Mudgett a écrit :
>> I've been using Grandstream phones for more than 10 years, but onl
> y
>> yesterday tried to use Early Dial... and I failed. What is needed
> on the
>> Asterisk side to reply 484 to INVITE? Phones are talking to chan_p
> jsip
>> on Asterisk-13.7.1.
>
>
>> Look into the Incomplete application.
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_In
c
> omplete
>
> Thanks for prompt answer Richard.
>
> Actually I had already tried the Incomplete application, but failed to
> add the "n" option, and this seems mandatory for SIP. I find the help
> text misleading : "NOTE: Most channel types need to be in Answer state
> in order to receive DTMF".
>
> This is my test dialplan:
>
> [earlydial] ; Test Early Dial
> exten => 1,1,Verbose(2,Incomplete 1 test)
> same => n,Incomplete(n)
>
> exten => _1X,1,Verbose(2,Incomplete 1X test)
> same => n,Incomplete(n)
>
> exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
> same => n,Playback(extension)
> same => n,SayDigits(${EXTEN})
> same => n,Hangup()
>
> It works, but seems a bit complicated: is this the correct way to use
> Incomplete ?
>
>
> Thanks,
>
> --
> _
> -- 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
>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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=hMTP
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Re: [asterisk-users] Grandstream Early Dial
Jean-Denis Girard
I have not used the Incomplete yet, but you might be able to do something like
this.
[earlydial]
exten => _.,1,Set(l_Extension = ${EXTEN})
exten => _.,n,Goto(${l_Extension},1)
exten => _.,n,Goto(noMatch,1)
exten => i,1,Goto(noMatch,1)
exten => noMatch,1, Incomplete(n)
exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
same => n,Playback(extension)
same => n,SayDigits(${EXTEN})
same => n,Hangup()
I wrote this in this message and have not tested this so use with caution.
There may be syntactical issues, but the concept might work for you.
Bryant
From: "Jean-Denis Girard"
Sent: Thursday, February 18, 2016 8:02 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Grandstream Early Dial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Le 18/02/2016 11:03, Richard Mudgett a écrit :
> I've been using Grandstream phones for more than 10 years, but onl
y
> yesterday tried to use Early Dial... and I failed. What is needed
on the
> Asterisk side to reply 484 to INVITE? Phones are talking to chan_p
jsip
> on Asterisk-13.7.1.
>
>
> Look into the Incomplete application.
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Inc
omplete
Thanks for prompt answer Richard.
Actually I had already tried the Incomplete application, but failed to
add the "n" option, and this seems mandatory for SIP. I find the help
text misleading : "NOTE: Most channel types need to be in Answer state
in order to receive DTMF".
This is my test dialplan:
[earlydial] ; Test Early Dial
exten => 1,1,Verbose(2,Incomplete 1 test)
same => n,Incomplete(n)
exten => _1X,1,Verbose(2,Incomplete 1X test)
same => n,Incomplete(n)
exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
same => n,Playback(extension)
same => n,SayDigits(${EXTEN})
same => n,Hangup()
It works, but seems a bit complicated: is this the correct way to use
Incomplete ?
Thanks,
- --
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
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Re: [asterisk-users] How to execute a macro after dial but before connect
Phillip Check out the b and B options one of them should do what you want. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Saint Michael" Sent: Friday, February 19, 2016 8:05 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] How to execute a macro after dial but before connect ?Dear friends: Is there a way to execute a macro or sub-routine after we send the invite before we receive anything like a 200 OK, 183, etc?? Philip -- _ -- 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] Voicemail issue on Grandstream GXP2000 phones
Richard Check both the DTMF settings, and the DialPlan string for account 3 on the phone. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Richard Schroeder" Sent: Tuesday, February 9, 2016 12:58 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] Voicemail issue on Grandstream GXP2000 phones Perhaps this is not limited to Grandstream GXP 2000 phones, but those are the phones we are using. Using FreePBX. Retrieving a voice message (*97) works fine from Line 1. Retrieving a voice message (*98) and picking the extension (Comedian mail) works fine from Line 1. From Line 3, it does not recognize the password. (*97 or *98). The extension is installed on Line 3. Retrieving Line 3's voice messages can only be done from Line 1 (on any extension on the PBX). Line 3 seems to work fine otherwise. Is this a limitation, or is it some kind of setup issue? I can't seem to find anything in the documentation for the phone or FreePBX related to this issue. Anyone? This is frustrating and I will be grateful for any help. Thank you! Richard -- Richard C. Schroeder rsch...@gmail.com rsch...@optonline.net 516-859-1129 - Cell -- _ -- 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 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API
Sonny We use a real-time database for adding pjsip users. If you want to do it from the pjsip.conf you would have to write to the file from a script of some sort and then trigger a reload. There is a real-time implementation for the extensions.conf as well. I personally use scripts for most of my dialplan, but in some cases I write to files included in my dialplan from a script and force a reload. To directly answer you question I do not believe there is an API baked into asterisk to update the pjsip.conf and extensions.conf directly from the dialplan. Thanks Bryant From: "Sonny Rajagopalan" Sent: Thursday, January 28, 2016 7:35 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API Hi, I am using Asterisk 13.6.0 and was wondering if I can programmatically add users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk server using API of some sort. Please do let me know. Thanks, Sonny. -- _ -- 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] PJSIP Stun/ICE
George Reloading transports is one critical part and it sounds like you are making headway on that. I have yet to be able to get transports to load from a real-time table using sorcery.conf If I would get the transports pulling from real-time as the (documentation says is possible but I have found no working examples yet) and then be able to reload any changes without forcing a compete asterisk restart. This would allow for a host of options for detecting and updating IP addresses. In the long run it would be nice to be able to tie some kind of stun support for updating the external media and signaling IP addresses. Thanks Bryant From: "George Joseph" Sent: Thursday, January 28, 2016 9:12 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] PJSIP Stun/ICE On Thu, Jan 28, 2016 at 6:58 PM, James Cloos wrote: > "AS" == A J Stiles writes: AS> If you are paying for a business-grade Internet connection, you AS> should get a static IP address -- or a block of them -- as AS> standard. Maybe you need to change your ISP? In some places (including here) static ip is not affordable. ?Please create a JIRA issue and let me know what the number is. I've just posted a patch for review that allows reloading transports from the command line.? I'd like to know what else you actually need. -- _ -- 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] PJSIP Stun/ICE
Joshua I look forward to improvements as time goes on with PJSIP. I have been trying all day to get the Transport objects to pull from a real-time table. The documentation says it is possible, but does not show any examples. I am hoping to have the Transports pulled from the table at asterisk startup and then add additional as necessary. Using reloads to make the new Transports available. I understand the limitation of not being able to change existing and can live with that for now. Do you know if there is anything special I have to do in the sorcery.conf to make the Transports pull from the real-time side of things. All my other tables are working. I disagree with the user that things PJSIP is worthless. There are some issues to work out long term, and documentation will get better over time as more of us work with it and contribute back. Thanks for all you have assisted with around PJSIP. Bryant From: "Joshua Colp" Sent: Tuesday, January 26, 2016 8:40 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] PJSIP Stun/ICE James Cloos wrote: >> "JC" == Joshua Colp writes: > > JC> This stems from PJSIP not being dynamic with transports (it > JC> doesn't like its environment changed to that degree while > JC> in use). I'm afraid if your IP changes you'd have to restart > JC> Asterisk when you are using PJSIP. > > Wow. > > I say this having voted for pjsip over the listed alternatives back when > the plan to depricate chan_sip was first floated: > > That should have excluded pj from the options. Which of course means > there were no reasonable options. PJSIP doesn't like changing existing transports, the NAT functionality is provided by the Asterisk implementation and can't be reloaded as a side effect due to the heavy handed restriction. With work it could be changed to allow the non low level things to be changed. What you can't do with PJSIP is create a UDP transport, reload, and have it removed. Once it's there it is there unless you restart. > > Can ari get around that bug? ARI is a REST interface to Asterisk, it doesn't have anything to do with this. > > Lack of full support for traversing nat makes pjsip worthless for a > large number of users. And the whole point of realtime is to have all > of the rt config fully dymanic. I disagree that it makes it worthless for a large number of users. It's only within the last few days that a few people have run into this particular issue where they have a public IP address that is changing a lot and PJSIP does not support changing it without a restart. If it were a huge sweeping issue we'd be seeing it more often. If it continues to show up a community member or us (heck maybe even myself in my spare time) may look into implementing it. > > If ari cannot avoid that limitation, chan_sip should get full ongoing > maintainance until pjsip is fixed. The support level for chan_sip has already been changed and was announced long ago. Patches will continue to be accepted for it and community members can support it. We (Digium) are putting our effort towards PJSIP. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] PJSIP Stun/ICE
Daniel Thank you for your response. I was considering this as well. I have a script that monitors the IP Address now. I was hoping to use the real-time transports table now that alembic creates. I am trying to figure out which pjsip module is responsible for the transports contexts as I need to now configure it in the sorcery.conf file. I thought it would be under the [res_pjsip] context, but it is not even trying to pull from my transports table when it is there. I am hoping someone will know what module it is in so I can move my configuration under the correct context. Thanks Bryant From: "Daniel Heckl" Sent: Tuesday, January 26, 2016 10:15 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] PJSIP Stun/ICE Bryant, I have the same problem with dynamic public IPs and PJSIP. What is your idea to solve the problem? My suggestion would be to write a script that monitors the change, pjsip.transports.conf updated and Asterisk restarts? Daniel > Am 26.01.2016 um 14:21 schrieb Joshua Colp : > > Bryant Zimmerman wrote: >> Joshua >> So once a transport is pulled from the transports table in realtime >> during asterisk startup it can't get any updates? >> Can a new transport be added to the table and the associated endpoints >> be updated to use the new transport, or are transport types only read at >> startup across the board? > > Transports can only be loaded at startup. This stems from PJSIP not being dynamic with transports (it doesn't like its environment changed to that degree while in use). I'm afraid if your IP changes you'd have to restart Asterisk when you are using PJSIP. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > > -- > _ > -- 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
[asterisk-users] PJSIP - Realtime - Transports module?
Does anyone know which module the type=transport loads under. I am trying to set up transports to load from a realtime table. I added the following under [res_pjsip] and it does not poll the associated database. [res_pjsip] transport=realtime,vap002_ps_transports We also set the associated values in extconfig.conf as well. My best guess is that transports are loaded under a different module's context. Anyone have an idea? Thanks Bryant -- _ -- 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] PJSIP Stun/ICE
Joshua So once a transport is pulled from the transports table in realtime during asterisk startup it can't get any updates? Can a new transport be added to the table and the associated endpoints be updated to use the new transport, or are transport types only read at startup across the board? Thanks Bryant From: "Joshua Colp" Sent: Tuesday, January 26, 2016 8:10 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] PJSIP Stun/ICE Bryant Zimmerman wrote: > Joshua > Since there is no automated way currently built in to update the > external signaling and media address information. > Does the realtime pjsip support having the transport contexts section > being pulled from a database table? > I was thinking a cron script updating the table and forcing a reload > each time an IP address changed might a workable solution. No, once loaded the transports can not be changed. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] PJSIP Stun/ICE
Joshua Since there is no automated way currently built in to update the external signaling and media address information. Does the realtime pjsip support having the transport contexts section being pulled from a database table? I was thinking a cron script updating the table and forcing a reload each time an IP address changed might a workable solution. Thanks Bryant From: "Joshua Colp" Sent: Tuesday, January 26, 2016 7:39 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] PJSIP Stun/ICE Bryant Zimmerman wrote: > I have an asterisk 13 server behind NAT on a dynamic IP Address. It is > running the PJSIP Stack > It is registering to another asterisk 13 server that is on a Static IP > outside of the firewall at a different location (also on the PJSIP Stack). > How do we implement STUN/ICE on the server behind the dynamic Address. > It does not appear to be registering properly without knowing the NAT > pubic address. When I manually add external_media_address and > external_signaling_address to the pjsipconfig registration seams to > work, but knowing that the IP could change really means I need some kind > of STUN/ICE similar to what we ran with chan_sip. > I can find limited documentation on this, and what I have found does not > show how to set a stun server to make the ice_support field work on an > endpoint. > Can anyone advise where I could find an answer to this. > Thanks in advance for any ideas you can offer. > Bryant The res_pjsip module does not currently support an auto-updating mechanism for the external signaling and media address information. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] PJSIP Stun/ICE
I have an asterisk 13 server behind NAT on a dynamic IP Address. It is running the PJSIP Stack It is registering to another asterisk 13 server that is on a Static IP outside of the firewall at a different location (also on the PJSIP Stack). How do we implement STUN/ICE on the server behind the dynamic Address. It does not appear to be registering properly without knowing the NAT pubic address. When I manually add external_media_address and external_signaling_address to the pjsipconfig registration seams to work, but knowing that the IP could change really means I need some kind of STUN/ICE similar to what we ran with chan_sip. I can find limited documentation on this, and what I have found does not show how to set a stun server to make the ice_support field work on an endpoint. Can anyone advise where I could find an answer to this. Thanks in advance for any ideas you can offer. Bryant -- _ -- 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] PJSIP NAT traversal.
I have two servers running pjsip they are both on NAT. The proxy has a static public address. I set the ;external_media_address=203.0.113.1 and ;external_signaling_address=203.0.113.1 to the actual IP address in the transport section on the proxy. The issue I am having is on the server with only a dynamic IP address. I can not figure out how to get ice support working so the public ip address is written into the registration.. The nat dynamic server is trying to register to the proxy. One issue I am seeing is only the private IP address are showing in the contact table on the proxies contact record. What could I be missing? Thanks Bryant -- _ -- 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] Using external RTP proxy for res_pjsip
Dmitrity What kind of volume are you running? You can use asterisk as a proxy if you set it up correctly. The choice would fall on the volume and the operational needs. To use an external proxy you would either need to register to the proxy or have a trusted IP to IP relationship. If your carrier allows for endpoint registration then you could attached your asterisk server directly, and would not need a proxy. If you have an IP for a proxy you could also do a NAT translation from that IP directly to the Asterisk server and negate the need for a proxy all together. As far as connecting to a proxy. You would need a pjsip endpoint either with a trusted IP or with a registration to your proxy server. As far as exactly what should go in your pjsip.conf that depends on your final implementation. You have not given enough detail of your network situation, and the reasons for a proxy to adequately advise you any deeper. Good luck. I hope some of the above is helpful to you. Bryant From: "Dmitriy Serov" Sent: Monday, November 2, 2015 9:10 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] Using external RTP proxy for res_pjsip The asterisk server has a permanent IP address, but the provider cannot ensure stable quality traffic for RTP. There is a desire to use an external server, the address of which shall be specified in the SDP, through which flowing media. I use asterisk 13.6 and res_pjsip. Prompt, please: 1. what software can be used on an external RTP proxy? 2. What settings need to be done in pjsip.conf to use this external RTP proxy? Preferably specifies the external RTP proxy to specify a specific endpoint, not globally. If only globally valid, the suit and the decision. I would be grateful for any clues. Dmitriy Serov. -- _ -- 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] Remote UNIX connection / disconnected.
Anyone know how to suppress the -- Remote UNIX connection / disconnected messages. I have a monitoring application that calls asterisk from the command line to verify some uptime stats. I would like to not have the console log the connections.. Any ideas are appreciated. Thanks Bryant -- _ -- 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] pjsip show xxxx like endpoint?
George and Mat Here is the link to the Jar Issue. https://issues.asterisk.org/jira/browse/ASTERISK-25477 Thanks Bryant From: "George Joseph" Sent: Sunday, October 18, 2015 10:17 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] pjsip show like endpoint? On Sun, Oct 18, 2015 at 5:07 PM, Matthew Jordan wrote: On Sun, Oct 18, 2015 at 12:39 PM, George Joseph wrote: > Did you open a Jira issue for this yet? I can actually work on this this > week. > I think it'd be pretty cool. George: want me to open an issue? Thanks Matt. Bryant said he'd do it tonight. -- _ -- 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] pjsip show xxxx like endpoint?
George, and Matthew I can open an issue later today, but if you want to do it that would be awesome as well. Please post the issue number back to this thread so I can follow it. Ideally the Like would work with all pjsip show commands so we can reduce the list and drill down just like we could with sip show commands This is a big missing for me right now and is really stopping me from going production along with the realtime performance issues already being talked about. Thanks for your assistance. I greatly appreciate it. Thanks Bryant From: "Matthew Jordan" Sent: Sunday, October 18, 2015 7:08 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] pjsip show like endpoint? On Sun, Oct 18, 2015 at 12:39 PM, George Joseph wrote: > Did you open a Jira issue for this yet? I can actually work on this this > week. > I think it'd be pretty cool. George: want me to open an issue? -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- 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] pjsip database error when using MS SQL via ODBC
I have a project that is requiring the use of MS SQL from asterisk. I get an error when the pjsip contact tries to update the contact table. [Oct 15 21:34:55] WARNING[3033]: res_odbc.c:649 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 22018: [FreeTDS][SQL Server]Conversion failed when converting the varchar value '3.00' to data type int. (101) The datatype in MySQL is integer and in MS SQL is integer. What could be the cause of this? Is it likely some kind of FeeTDS conversion issue? If I change the MS SQL type to double the error goes away, but I am unsure of the long term issues associated with this. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] pjsip show xxxx like endpoint?
Is there a way to limit the items returned by pjsip show [type] using like chan_sip allowed for sip show peers like , but I can't seem to figure out how to lookup or limit my returns with pjsip Thanks Bryant -- _ -- 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] pjsip realtime registrations not pulling from ODBC
From: "Joshua Colp" Sent: Monday, October 5, 2015 9:20 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-05 10:15 AM, Bryant Zimmerman wrote: > > -- > I am working on a step by step and some internal documentation on pjsip. > Where would the best place to post this information be. I am willing to > share what I have it may help others in the same boat I was in. > I have been running pjsip internally for several 6 plus months and still > hit config snags occasionally, but our config process is almost complete > and is going production by the end of the month. There's a section on the wiki[1] for configuring PJSIP. You can take a look and see what is missing/could be improved. If you'd like you can add a comment to an initial page and if your suggestions look good then Rusty can provide you wiki edit access. [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- Joshua That sounds good. I will do this as soon as we know our documentation is good. Thanks Bryant -- _ -- 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] pjsip realtime registrations not pulling from ODBC
From: "Ryan, Travis" Sent: Monday, October 5, 2015 8:20 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC Ah ok, thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Monday, October 05, 2015 8:20 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-05 09:16 AM, Ryan, Travis wrote: [snip] > > > So should anyone using realtime PJSIP be using the registrations line? Even if it's not used for any trunking? A registrations line in sorcery.conf for res_pjsip would do absolutely nothing. If you put it under res_pjsip_outbound_registration and have no outbound registrations it will execute some queries against your database but otherwise do nothing. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- I am working on a step by step and some internal documentation on pjsip. Where would the best place to post this information be. I am willing to share what I have it may help others in the same boat I was in. I have been running pjsip internally for several 6 plus months and still hit config snags occasionally, but our config process is almost complete and is going production by the end of the month. Bryant -- _ -- 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 Qualify to pjsip
I am running a pjsip test between two servers one running pjsip and one running chan_sip The chan_sip side is sending requests based on qualify=yes. The pjsip side is showing notices.. Exp ?[Oct 4 18:09:02] NOTICE[5982]: res_pjsip/pjsip_distributor.c:347 log_unidentified_request: Request from '"asterisk" ' failed for 'xxx.xxx.xxx.xxx:5060' (callid: 2d65aa1a2f162b075486d21b661c9...@xxx.xxx.xxx.xxx:5060) - No matching endpoint found Why is asterisk chan_sip sending it's qualify requests as from sip:aster...@xxx.xxx.xxx.xxx? Is there a way to get the chan_sip side to send it's qualify requests from the actual registered peers so these messages would not look like requests being sent from an unauthorized endpoint? Is there an easy way to supress these notices if there is no way to shut them down. Thanks Bryant -- _ -- 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] pjsip realtime registrations not pulling from ODBC
From: "Joshua Colp" Sent: Sunday, October 4, 2015 12:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:42 PM, Bryant Zimmerman wrote: > > *From*: "Joshua Colp" > *Sent*: Sunday, October 4, 2015 12:12 PM > *To*: asterisk-users@lists.digium.com > *Subject*: Re: [asterisk-users] pjsip realtime registrations not pulling > from ODBC > On 15-10-04 01:09 PM, Bryant Zimmerman wrote: >> -- >> Joshua >> Thanks for your reply. It thought the same thing, but when I change the >> line in the corcery.conf to: >> registration=realtime,px1_ps_registrations >> Asterisk crashes and won't start. Here is what the log loop. >> [Oct 4 16:04:18] WARNING[64823] config_options.c: Cannot update type >> 'registration' in module 'res_pjsip' because it has no existing >> documentation! >> If I switch to "registrations=realtime,px1_ps_registrations" the error >> stops, but I get now calls from the px1_ps_registrations table from the >> database. >> What could be missing? > > Outbound registrations are done in res_pjsip_outbound_registration, as a > result the registration= needs to be in a section for that module instead. > > -- > Joshua > That seems to have been the issue. Is there a documentation page out > there that highlights which options goes under which modules. > I have not run across this yet and am wondering if I am going to bump > into any more that need to be pushed under their own config context. I don't think there's a page that describes it unfortunately. > Also is there a trick to what should be used in the client_uri field to > make the connection? > I am trying to connect to a sip vendor and I am trying to use > sipacco...@venderhostname.vendordomain.net? > Now that the registration table is coming up it is stating I have an > invalid client URI.. I put the same thing in for a text based > registration and it worked. You need to put sip: in front to make it a valid SIP URI. -- Joshua Much thanks for all your direction today. I am sending good karma your way. Thank You Thank You Bryant -- _ -- 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] pjsip realtime registrations not pulling from ODBC
From: "Joshua Colp" Sent: Sunday, October 4, 2015 12:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:09 PM, Bryant Zimmerman wrote: > -- > Joshua > Thanks for your reply. It thought the same thing, but when I change the > line in the corcery.conf to: > registration=realtime,px1_ps_registrations > Asterisk crashes and won't start. Here is what the log loop. > [Oct 4 16:04:18] WARNING[64823] config_options.c: Cannot update type > 'registration' in module 'res_pjsip' because it has no existing > documentation! > If I switch to "registrations=realtime,px1_ps_registrations" the error > stops, but I get now calls from the px1_ps_registrations table from the > database. > What could be missing? Outbound registrations are done in res_pjsip_outbound_registration, as a result the registration= needs to be in a section for that module instead. -- Joshua That seems to have been the issue. Is there a documentation page out there that highlights which options goes under which modules. I have not run across this yet and am wondering if I am going to bump into any more that need to be pushed under their own config context. Also is there a trick to what should be used in the client_uri field to make the connection? I am trying to connect to a sip vendor and I am trying to use sipacco...@venderhostname.vendordomain.net? Now that the registration table is coming up it is stating I have an invalid client URI.. I put the same thing in for a text based registration and it worked. Thank you for your assistance. Bryant -- _ -- 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] pjsip realtime registrations not pulling from ODBC
On 15-10-04 09:54 AM, Bryant Zimmerman wrote: > I have a pjsip install that is not pulling it's realtime registrations > from an ODBC database. > When I have them directly pull from a MySQL database everything seems to > work. > I am having trouble finding a requirements document for the pjsip > realtime registrations. > Is there some kind of special config that has to be used to trigger the > connection for realtime registrations over ODBC? > My realtime connections to aors, auths,contacts, and endpoints via ODBC > are working as expected. > Any ideas are appreciated. > Asterisk v 13.5.0 > Registrations line from sorcery.conf > /registrations=realtime,px1_ps_registrations/ This should be: registration=realtime,px1_ps_registrations > Line for database from extconfig.conf > /px1_ps_registrations => odbc,pjsipRealtime/ > pjsip show registration > show no return records - even though there are records in the database > table. > Thanks > > Bryant > > -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- Joshua Thanks for your reply. It thought the same thing, but when I change the line in the corcery.conf to: registration=realtime,px1_ps_registrations Asterisk crashes and won't start. Here is what the log loop. [Oct 4 16:04:18] WARNING[64823] config_options.c: Cannot update type 'registration' in module 'res_pjsip' because it has no existing documentation! If I switch to "registrations=realtime,px1_ps_registrations" the error stops, but I get now calls from the px1_ps_registrations table from the database. What could be missing? Thanks Bryant -- _ -- 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] pjsip realtime registrations not pulling from ODBC
I have a pjsip install that is not pulling it's realtime registrations from an ODBC database. When I have them directly pull from a MySQL database everything seems to work. I am having trouble finding a requirements document for the pjsip realtime registrations. Is there some kind of special config that has to be used to trigger the connection for realtime registrations over ODBC? My realtime connections to aors, auths,contacts, and endpoints via ODBC are working as expected. Any ideas are appreciated. Asterisk v 13.5.0 Registrations line from sorcery.conf registrations=realtime,px1_ps_registrations Line for database from extconfig.conf px1_ps_registrations => odbc,pjsipRealtime pjsip show registration show no return records - even though there are records in the database table. Thanks Bryant -- _ -- 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 AMI events filtering
Sam Based on my experience you need to write a middle tier that has what you want exposed to the users.. AMI was not really designed to offer direct multi-tenant access. That is for your middle tier to handle. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Sam Basan" Sent: Thursday, September 17, 2015 7:21 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] Asterisk AMI events filtering Hi folks, I have one server with multiple companies (multi-tenant). >From AMI I get all events of all extensions so any one that connect can see other extensions, from different company (context). How can I limit specific user to get just specific context? Sam -- _ -- 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 SMS
On Friday 10 Jul 2015, Thyda ENG wrote: > Dear Sir, > > Does the asterisk support SMS feature ? > If it does how can we config that ? > I am waiting for your reply,Thank. Thyda Yes asterisk supports SMS on both cell card and sip trunk. Checkout this link as a starting point. http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] Distributed Device States - Best Option
We have used AIS for disturbed Device State in the past, BLF and MWI, We are in the process of an update on one of our clustered systems, We are looking at XMPP and I found a few discussions on a Corosync with has OpenAIS built in. My question is which should I be looking at to replace my current AIS option I currently have. XMPP or Corosync? It looks like the Corosync is just the AIS option more nicely packaged. Is XMPP a better solution as I grow my network? Are there down sides to XMPP that AIS/Corosync does better... Can anyone recommend where I can find some up to date documentation that would cover up through Asterisk 13 on Distributed Device State. Thanks for any feed back. Bryant -- _ -- 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] adding area code
Thanks for your reply,
[globals]
AREACODE=381
[outbound]
exten => _9XX,1,Dial(SIP/SIP-Provider/1${AREACODE}${EXTEN-1},80)
did not work for me, any ideas?
Thanks,
On 04/27/2015 01:59 PM, Phil Reynolds wrote:
On 27 April 2015 21:32:42 BST, Motty Cruz wrote:
>Hello,
>
>I would like to add area code if clients dial 7 digits, it that
>possible? currently clients dial prefix 9 plus local number, however my
>
>SIP provider is requiring to dial 10 digits. is it possible to add area
>
>code?r
Quite simple - you need to match on NXXX and when passing it to the SIP
provider, present ${AREACODE}${EXTEN}, having first defined AREACODE in
[globals].
--
Sent from my Android device with K-9 Mail. Please excuse my brevity.
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Re: [asterisk-users] adding area code
Motty
Yes
From your dial plan accept 9 + 7 digits then concat your dialed number
together with your areacode.
This s a brief example.
exten => _9XXX,1,Set(l_HomeAreaCode=555)
exten => _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) ;; This
line should combine your area code and the last 7 digits of your dialed
phone number
exten => _9XXX,n,Dial(SIP/${dialnumber},35)
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: "Motty Cruz"
Sent: Monday, April 27, 2015 4:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] adding area code
Hello,
I would like to add area code if clients dial 7 digits, it that
possible? currently clients dial prefix 9 plus local number, however my
SIP provider is requiring to dial 10 digits. is it possible to add area
code?
Thanks,
Motty
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Re: [asterisk-users] FXO advice
Alejandro All of the Grandstream devices can be remote provisioned if you know what you are doing. Bryant From: "Alejandro" Sent: Wednesday, April 15, 2015 4:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FXO advice Hi All, I'll like to know if exist some Basic FXO that support some type of automatic provisioning of configuration. Our idea is avoid the users need to go into WebPage and setup our SIP gateway. Some advice or recommendation? Thanks Alejandro -- _ -- 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] Grandstream GXP2140
We use a lot of GXP21x phones. We have had issues with the GXP-2140 when using the side car as BLF's. The device becomes sluggish after about 45 days of operation a reboot solves the issues. This has been reported but not resolved as of yet.. If you are not using the side car the issue does not seem to pop up.. We just put the latest firmware on the phone but it does not say it fixes the issue so we are not expecting it to.. If you don't need the side car they are a good phone. Thanks Bryant From: dsi...@hcmr.gr Sent: Wednesday, April 15, 2015 3:12 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Grandstream GXP2140 I'm working with GXP2130. About 12 phone on gigabit with PC after phone. With Vlans on CISCO switch is stable and not so difficult. This configuration running without problems since July 2013. Quoting jg : >> I have a customer looking to deploy about 20 Grandstream GXP2140 >> phones. Normally they would deploy Yealink brand phones but they >> want a phone with gigabit pass through and the Yealinks with >> gigabit are too expensive for their budget. >> >> >> Does anyone on the list have experience with the GXP2140? Is it a >> reliable phone? Does anyone have recommendations for other phones >> with gigabit pass through? >> >> > I'd be generally careful with the second ethernet connection. One > should look at the chipset of the phone. I had pretty bad > experiences with somewhat older TI based phones, regardless of the > manufacturer. The problems became apparent in mixed environments, > where some connections were gigabit and others not. It can be a > nightmare, if you have to offer support. > > The best bet is to buy one, and check the performance of the > connections. I use some GrandStream products myself and the product > quality is now much better compared to a couple of years ago. > > jg > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > > D. Sidirokastritis NOC HCMR-Crete tel. 2810-337709 Hellenic Center for Marine Research This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- 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] Fidelio protocol and Mitel protocol
Does anyone know anything about the Fidelio and Mitel protocol for hotel / motel? Are these industry standards or proprietary formats? Are there open standards for communication with Hotel management software's that could be used in conjunction with a custom asterisk deployment? Thanks Bryant -- _ -- 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] Unstable phone connection
D'Arcy J.M. Cain If the device is registering and then dropping there are several usual items. The router may be closing the ports on the device. The router may have a AGL SIP helper that is causing issues. Make sure that the device is sending out keep alive packets. Shut down any AGL helpers on the router. Make sure that the site is not double NATing Try using a stun server and see if that helps at all. Watch you console on your sip serer to see how long the device runs before losing connection. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "D'Arcy J.M. Cain" Sent: Thursday, March 12, 2015 2:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unstable phone connection This is driving me to distraction. I have a switch with multiple clients who are all working fine except for one and I can't figure out what makes them different. I have tried every NAT setting in the ATA (SPA112 ATA with 2 x FXS, 1 x LAN), stun server on and off, different sip ports, different RTP ports and it still fails. I have left the location with it working only to have it fail later. He always gets registered but when a call is sent it doesn't respond so the caller hears no ring and the phone does not ring. Yesterday he mentioned that when the phone is working the WiFi slows down significantly. No idea why or if it is related. He has a radio station streaming music. I wondered if that might be interfering. That's why I tried changing the SIP port and the RTP ports but that didn't seem to help. It smells like a network problem to me but I am running the same ADSL device here and other clients are working behind a NAT gateway so I am at a loss as to what might be wrong. Could it be the streaming? Cheers. -- 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 -- _ -- 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] GXP 1405 and asterisk
SIPAddHeader(Alert-Info:\;info=ring3) In the phone config add the value "ring3" and select Account # / Call Settings / Match Incoming Caller ID (Matching Rule) In the first rule place the word ring3 and select your ring tone. This will cause the selected ringtone to be used when calls with the info value of ring3 is matched Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "ricky gutierrez" Sent: Thursday, March 12, 2015 2:42 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] GXP 1405 and asterisk Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com -- _ -- 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] switching from SIP to Skype..or not
Hey all We have been working with SIP for years. It has the potential to be better than Skype. It is really all in the implementation. Not all SIP soft clients are equal nor are the networks and computers they are running on. I will not bash Skype. We have tested it and in most cases choose not to use it. It has it's place and is good for the user that meets it's specific target demographic. SIP is a sold communications protocol that can communication with codecs of differ audio and video quality levels, and supports industry standard software and hardware endpoints. With SIP you get to choose how good your quality is. With Skype Microsoft does. It comes down to what do you want to achieve, how much resource do you want to put in to it, and are you committed to a bit more work for a lot more options and better quality, or do you want a quick and easy solution with differing limits. Both solutions have their place. To me SIP vs Skype is like complaining apples and carrots do you want fruit or veggies you get to choose. You can choose to agree or disagree with my statements. I hope they are useful to some. Thanks Bryant From: "Ron Wheeler" Sent: Thursday, March 12, 2015 9:40 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] switching from SIP to Skype..or not Your characterization may be true but Skype works much better than SIP when it comes to sound quality. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. Ron On 12/03/2015 9:26 AM, A J Stiles wrote: > On Thursday 12 Mar 2015, Thufir wrote: >> I'm testing Asterisk at home, crummy connection. Skype works fine for >> me, but every SIP client, even without using Asterisk, fails to connect. >> That's ok. >> >> Is swapping out SIP for Skype a big deal? > Stay away from Skype! It is a toxic, proprietary product. The lack of > interoperability by design is the antithesis of what a telecommunication > system should be about -- and the extent to which they have gone to thwart any > attempt at interoperability is truly shocking. > > For connecting two Asterisk installations to each other over the Internet, IAX > is better than SIP -- that's what it was designed for. > -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] func_odbc 123
Thurfir We use a combination of func_odbc calls to drive static dialplan. Be aware that there is currently a bug if you use ODBC with MySQL, and your primary database is offline The system does not properly roll to backup database servers. This often causes asterisk to lock up requiring a restart. Thanks Bryant Zimmerman (ZK Tech Inc.) From: "Thufir" Sent: Tuesday, March 10, 2015 4:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] func_odbc 123 with func_odbc, in the definitive asterisk guide, they were suggesting the possibility that part, or perhaps all of, the dialplan could be written as SQL statement!? First off, that sounds like a good idea to me, but the tone of the authors was suggesting not so much, but that it was a personal preference. >From a naive perspective, why SQL statements at all? Why not just database config and data instead? thanks, Thufir -- _ -- 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] New Asterisk build
John I will have to get one of these and give this a try. Thanks for sharing. Thanks Bryant Zimmerman (Grand Dial Communications, a ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "John Novack SCII" Sent: Friday, March 6, 2015 3:37 PM To: "Ira" , "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] New Asterisk build Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux ( instructions at AstLinux.org ) Move your Digium card and your confs , fix up any differences from your older version of Asterisk to the fairly current version 11 currently available with AstLinux. Use the GUI to edit and mage the system, as AstLinux has a somewhat different directory structure than you may be familiar with You should be up and running in a couple of hours, have a low power < 20 watts, fanless flash based system that will just work in a real case. The Pi is OK for a playtoy and some testing, but I much prefer the HP thin clients for a robust installation. I assume you are not doing any fancy call center or heavy database work. For a home or home office it is a really good solution. AstLinux is also used with other embedded installations on computers with multiple Ethernet ports, acting as router and firewall in addition. I prefer the component solution personally, which makes the HP thin clients the way to go. John Novack I have built more than 30 of these systems on various HP Thin Clients, used off of eBay with no failures Ira wrote: > Hello Asterisk, > > Back in 2009 I built a small Intel Atom based computer running > Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs > line and six or so SIP numbers. So basically no load. I'm > feeling like it's time to build another machine. It's probably > silly, but it's been six years and I can't upgrade the OS > which is falling behind. I'd likely just put it on a Raspberry > Pi or something like that, but I need the one POTS line and > all I have for that at the moment is a Digium card for a PCI > slot. > > Are there any current thoughts on this? > > -- Ira > > -- Dog is my Co-Pilot -- _ -- 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] New Asterisk build
Iran For the kind of loads and low cost you are talking with 2 FXO, 2FXS and SIP the Grandstream UMC6102 is low power feature rich and easy to maintain. Check it out - http://www.grandstream.com/index.php/products/ip-voice-telephony/ip-pbx-solu tions/ucm61xx If you do choose to use the UMC61xx the grandstream phones auto-provision with it well, but it works with any complaint SIP phone. If you do want to go with an asterisk home brew. You could use a Grandstream GXW4104 (4 FXO) for your POTS line. It is a FXO gateway that would register as a SIP endpoint. (You could look at the HT503 which has one FXO port, but I find them to be less reliable then the GXW4104). The nice thing about using gateways is there are no drivers to load on your asterisk build as the gateway is just a SIP endpoint. I have built asterisk test systems on raspberry pi Rev B and have not been happy with their performance even in light loads. The new version 2 B looks like it might be better, In ether case the Gateways would be a good way to go to connect your lines. Watch your SD card speeds slow cards really gave me a lot of issues. Especially when you had someone leaving a voicemail and someone else was trying to listing to an IVR prompt, multiple users reading and writing at the same time just really have not worked well. We hooked up a SSD via USB and put our prompts and voicemail on it and it was a bit better still limited to USB2 speeds, but that increased the cost. The UMC6102 is the best value as buy the time you purchase a gateway, system and spend time loading it is hard to beat the price point and you can get support on it from Grandstream or a reseller. (To be open I am a Grandstream reseller, I am offering these recommending as they are good options. There are several other low cost asterisk like PBX's out there as well, Allo and several others, but I know the GS options work) Good Luck and I hope this info helps. Thanks Bryant Zimmerman (Grand Dial Communications, a ZK Tech Inc.) P.S. Glen's post also offers some good points as well. -- _ -- 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] Investigating international calls fraud
If you have not done so contact the carrier immediately. Report the fraud. Have them disable international on the account until you have your security issues addressed. Ask them to pull call logs containing Source and destination IP address. for the fraud calls. If you are sure they came from your systems IP address then verify the systems does not have any unauthorized registered endpoints. Verify the source and destination of the calls from any internal logs. If the calls came from your IP then you are likely on the hook for the calls. If they came from another IP that you don't own then if you can prove you had no access to that IP than there is some hope the carrier may work with you. Thanks Bryant -- _ -- 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] Return SIP 401 on hangup
I am having and issue I hope someone can help with.. I have calls that often come in that need to be blocked. We wish to do this without answering the call. The issue is our carriers have fail over servers and will try sending the call from each when we block the call. If we send a hangup with a Sip 401 they will stop the route advance on their end. The issues is we have been sending a hangup/cause code 21 (Call rejected) But they are receiving a 403 Forbidden.. Is there any hangup code that we can send that will reply a 401.. We see the 401 on the inbound is converted to a cause code 21 but we do not see any cause codes listed to send a 401 out. Please advise as this a becoming an issue for us as we have multiple vendors expecting a 401 not a 403 Any assistance is appreciated. Thank you. Bryant -- _ -- 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] Confbridge
I am doing dynamic conference bridges using confbridge in asterisk 11. Is there a way to toggle off an on recording of an ongoing conference call I have figured out how to record a conference if it is turned on when someone enters. Also I have noticed that when setting music_on_hold_class dynamically it does not override what is set on the channel. exten => s,n,Set(CONFBRIDGE(user,music_on_hold_class)=latin) Does anyone have any ideas on how I might fix this as well? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] Astricom 2014 presentations
On 10/29/2014 05:50 AM, Bogdan Cristea wrote: > Hi > > Will the presentations made at Astricom 2014 be made public as recorded videos ? > > thanks > Bogdan I'll second the request for that, and also ask if the sessions on Kamailio will be similarly available. Cheers, j That would be awesome if they chose to do this. Bryant -- _ -- 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] unidata incom ICW-1000G - On asterisk
I am trying to use an ICW-1000G wireless handset connected to an asterisk server remotely The user is working from an offsite location and it appears that the device is not sending out keep-alives or stun. The manufacture is not being of assistance at all. I am wondering if anyone has worked with these units or has any ideas of what I could do to make them work. Anyone have a better alternative. The units must support wifi, be able to clip on belt and support an external headset (wired is fine) Battery life should support all day use in a standard biz env. Work thru a firewall from offsite. The ICW-1000G supports all of these requirements except the Work thru a firewall from off-site. If the MFG can't get a fix or someone does not have any ideas I would recommend people stay away from the ICW-1000G if you have to use them off-site or for hosted connections. Thanks Bryant -- _ -- 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] AGI scripts - delay issue.
Hey All We have several AGI scripts that access databases. These work well most of the time. The issue we are having is that on rare occasion our script must fail to a backup database server. When this occurs it may take up to two seconds to do so. The issue is when there is this delay the script loses access to read global channel variable values only after the delay. This is driving me crazy is there some kind of AGI timeout issue or bug that could be causing this. Thanks Bryant -- _ -- 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] Dynamic Parking Lots. Music on Hold Class
How can we set the music on hold class using the Dynamic Parking lots? The variables set the PARKINGLOT, PARKINGDYNAMIC, PARKINGDYNPOS,PARKINGEXT,PARKINGDYNCONTEXT I can't find a PARKINGMOH variable. This is becoming a big issue. We are using the current release 11. version We have to be able to set the MOH dynamically I just can't find the mechanism. Any ideas? Thanks Bryant -- _ -- 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] AGI script VERBOSE cmd
Hey all Please disregard my question. I was looking for the word Verbose to show up. I was just being dense. There was no real issue it is working just different than what I was expecting. Thanks Bryant From: "Bryant Zimmerman" Sent: Friday, June 27, 2014 11:25 AM I am working on an AGI script and all is going well except I can not get any of my "VERBOSE" commands to display. Is there any undocumented reason for this to occur? I am able to set variables, call other commands ect.. I am sending my verbose command in the following format (VERBOSE "Message to send" 4) Any ideas what I might be doing incorrect? Thanks Bryant -- _ -- 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] AGI script VERBOSE cmd
I am working on an AGI script and all is going well except I can not get any of my "VERBOSE" commands to display. Is there any undocumented reason for this to occur? I am able to set variables, call other commands ect.. I am sending my verbose command in the following format (VERBOSE "Message to send" 4) Any ideas what I might be doing incorrect? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] Live Recording on the Storage Server?
A simple way that we use to do the move is to create a cron job that looks for a .move file. It has the same name as the recorded file. asterisk writes the .move file which is just a text file with some stats in it. The .move file is written from the dial plan at the end of the recording. In the exten = h we write a .delete file for an abandon call. The cron then processes the .move and .delete files at a given interval. We actually write special instructions into our .move files that the cron parses and can then act accordingly. So we have a single smart cron job handling moves for each type of task. In some cases our .delete files are processed as moves to an abandon cache for recovery if a customer did not intend to abandon it. The sky's the limit on how complex you want to make it, but in the long run it is fairly simple and it just works. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Chris Bagnall" Sent: Thursday, April 17, 2014 11:32 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Live Recording on the Storage Server? On 17 Apr 2014, at 16:14, Paul Belanger wrote: >> hi. I would not do that due to network issues. >> My approach is to record everything locally and every hour or so to move >> everything to a storage. > +1 save yourself the headache and do this. I'll add another +1 to this. I've never been able to get multi-channel recording (even 3 or 4 channels) working reliably over an NFS link to another server. I suspect, with some tweaking of nfs options it might be possible, but if it ain't broke. Just a cautionary note if you do use a cron job to move recordings to a storage device at regular intervals: make sure you use lsof or similar to check the recordings aren't actually open by asterisk at the time, otherwise interesting things will happen. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] func_odbc
Hi All Anyone know how to do include files with func_odbc.conf? I now have several pages of functions in my func_odbc.conf and it is getting harder to maintain it. I would like to break them up into files by category. The standard method of using the #include does not seem to work . Ideas are appreciated. Bryant -- _ -- 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 SIP server on windows
Ruddy If you can target windows 8.0 pro or 8.1 pro you can ship a Hyper-V image. (This same image would work with Hyper-V on, Hyper-V Server 2012/2012r2 and Windows Server 2012/2012r2. You would need to write some kind of configuration editor or documentation to customize the image to the users network environment. You could automate the entire process if you were so inclined. You can fully automate the Hyper-V side of things using PowerShell or Dot.Net The Linux side of things can be automated in a number of ways. We personally wrote a windows program that collects information from the user and posts it to our databases. The default image then has a script that pulls the info down (images uses DHCP to start) and re-writes the asterisk configs. This process is not a small task but if you have the time and budge it can work very well. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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 SIP server on windows
On Wednesday 04 December 2013, Ruddy Gbaguidi wrote: > Hi all, > > I need to build an application that will be an SIP server program that will > run on Linux and Windows. > > The sip server need only some features such as be able to : > > - Register sip endpoints > > - Answer a call and play a local file > > - Make a dial from one channel to another. > > > > I know asterisk can be stripped to exactly fit my needs. I would like to > know if there is a way to build it on windows after it has been stripped. > > Or do I have other alternatives out there ? Ruddy If you can use windows 8.1 Pro 64bit. You can use hyper-v and run a virtual linux machine (Or the Free Hyper-V Server 2012 R2), VM Ware also works well. Load asterisk on that and you are set. This is how we run it at a few very small customers as well as my development machines and it works great. Best linux builds for Hyper-V we currently have found to be Ubuntu and Suse. As both a windows and linux guy I have to concur that loading Asterisk on windows directly is like putting a V8 on a moped. You may get there, but it won't be pretty; It's a lot of work, and it would be hell to maintain. (We do not trust it for production applications) In all seriousness I have a Asterisk build running on windows and it is stable but it is a lot of work to get it there and since it is not maintained by the community it is a full task to keep it up to date. We use it for in process testing of code that we develop with MS visual studio. If not for that I would not bother with Asterisk on Windows there would be no value in it. Especially since the current version of Asterisk now works so well in virtual environments. Good luck Bryant -- _ -- 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] Voicemail greeting playback issues?
From: "Doug Lytle" Sent: Monday, November 25, 2013 6:25 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Voicemail greeting playback issues? Bryant Zimmerman wrote: > Hey all > > I believe I found the bug in Asterisk 11.xxx If someone can help me > verify it. Actually, I wouldn't consider it a bug. I've know for years that you need to answer a channel before you play back audio or strange things can and will happen. Doug -- Doug The real issue here is that issuing an Answer() just before does not seem to solve the problem. To work around the issue I have to either put a Wait(1) or Dial() some extensions first. It is presenting like if you drop into the Voicemail() command too fast during call setup that you have issues. This did not occur in 1.8.x. I would be ok if just issuing an Answer() would resolve it as this would be normal, but having to slow down the dial plan seems off. Thanks Bryant -- _ -- 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] Voicemail greeting playback issues?
From: "Bryant Zimmerman"
Sent: Monday, November 25, 2013 2:49 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Voicemail greeting playback issues?
From: "Doug Lytle"
Sent: Monday, November 25, 2013 2:01 PM
To: brya...@zktech.com, "Asterisk Users Mailing List - Non-Commercial
Discussion"
Subject: Re: [asterisk-users] Voicemail greeting playback issues?
>> Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue
goes away.
I don't see this under 11.5.1
Doug
---
Doug
Thank you for your response. It is good to hear that you are not having the
issue.
It gives me hope that there is a way to resolve this quickly.
Do you have an thing special around your voicemail configuration? We
started with the 11.xx sample config and mapped our settings from 1.8.x.
Both our 11.xx and 1.8.x systems are running on the same virtual server.
Both are reading and writing audio and vm files to and from the local
storage. I forced off g729 to ensure that it was not causing the issues.
Do you know of any way to force a higher level of debugging to see why the
voicemail application would be having an issue?
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
Hey all
I believe I found the bug in Asterisk 11.xxx If someone can help me verify
it.
My voice mail test scripts do not answer or wait they just drop you into
the voicemail box.
It appears that something with Asterisk 11.xx is causing the voicemail()
command to drop in and ether not play or mess up the prompts. If you have
not given it at least one second in the channel before passing it to the
voicemail() command.
If you throw a wiat(1) just before the voicemail() command the prompts play
correctly. So if you have rung extensions using dial() before going to
voicemail that appears to be enough time.
If you place an inbound call directly to voicemail() with no pause then you
have an issue.
Example Broken:
exten => _9XXX,1,Set(l_VMExt=${EXTEN:1})
exten => _9XXX,n,MailboxExists(${l_VMExt}@${siteVMContext})
exten => _9XXX,n,GotoIf($["${VMBOXEXISTSSTATUS}"="FAILED"]?doHangup)
exten => _9XXX,n,Voicemail(${l_VMExt}@${siteVMContext},u)
exten => _9XXX,n(doHangup),NoOp(Issue 9XXX Hangup)
exten => _9XXX,n,Hangup()
Example Works:
exten => _9XXX,1,Set(l_VMExt=${EXTEN:1})
exten => _9XXX,n,MailboxExists(${l_VMExt}@${siteVMContext})
exten => _9XXX,n,GotoIf($["${VMBOXEXISTSSTATUS}"="FAILED"]?doHangup)
exten => _9XXX,n,Wait(1)
exten => _9XXX,n,Voicemail(${l_VMExt}@${siteVMContext},u)
exten => _9XXX,n(doHangup),NoOp(Issue 9XXX Hangup)
exten => _9XXX,n,Hangup()
The code that is broken with Asterisk 11.xx worked in Asterisk 1.8.x
Can anyone confirm this?
Thanks
Bryant Zimmerman()
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Re: [asterisk-users] Voicemail greeting playback issues?
From: "Doug Lytle" Sent: Monday, November 25, 2013 2:01 PM To: brya...@zktech.com, "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Voicemail greeting playback issues? >> Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue goes away. I don't see this under 11.5.1 Doug --- Doug Thank you for your response. It is good to hear that you are not having the issue. It gives me hope that there is a way to resolve this quickly. Do you have an thing special around your voicemail configuration? We started with the 11.xx sample config and mapped our settings from 1.8.x. Both our 11.xx and 1.8.x systems are running on the same virtual server. Both are reading and writing audio and vm files to and from the local storage. I forced off g729 to ensure that it was not causing the issues. Do you know of any way to force a higher level of debugging to see why the voicemail application would be having an issue? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] Voicemail greeting playback issues?
Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked and the greeting files are there and play back from the voicemail ivr. If no greeting is there it just plays "The Pers.. beep" Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue goes away. Any Ideas? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] 11.6 voicemail message cropped off?
Update When no greeting is recorded the default you have reached ext # greeting is cropped. When there is a greeting it is just ignored and not played at all. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Bryant Zimmerman" Sent: Saturday, November 23, 2013 8:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 11.6 voicemail message cropped off? Hey all I am running 11.6 and when a caller is sent to vociemail the greeting is cropped off and the beep occurs quickly. Incoming calls are g729 and this occurs where it is using the default greeting or a user provided greeting. I really want to go production with this are there any ideas what could cause an issue like this we have never seen it in 1.4 - 1.8 Bryant -- _ -- 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] 11.6 voicemail message cropped off?
Hey all I am running 11.6 and when a caller is sent to vociemail the greeting is cropped off and the beep occurs quickly. Incoming calls are g729 and this occurs where it is using the default greeting or a user provided greeting. I really want to go production with this are there any ideas what could cause an issue like this we have never seen it in 1.4 - 1.8 Bryant -- _ -- 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] Movistar sip Mexico
Can you funnel them through a specific inbound dial context. Then force a re-invite to g729? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Damian Gonzalez" Sent: Thursday, November 21, 2013 8:25 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Movistar sip Mexico Any posible solution? On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner wrote: It is possible that Asterisk requires an rtpmap even for static payload types (I'm not sure about this). The INVITE from your provider omits rtpmap for payload type 18 (G729) which is perfectly valid. On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez wrote: Hello, Thanks for the quickly response. I have only G729 in the peer but I have t38pt_udptl= yes If I put t38pt_udptl=no , asterisk reject the call with 488 code. The problem is that Movistar send T38 codec in all calls and I need ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have only T38 I have to negociate a fax call. Thanks. On Wed, Nov 20, 2013 at 4:46 PM, Alyed wrote: Think you only need to make sure you have in your sip.conf file these configs: [your-device-name] . . disallow=all allow=g729 . . Alyed 2013/11/20 Damian Gonzalez Hello, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 6372 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy How can I ignore T38 and use only G729 for this call?. Thanks for your help. Damian -- -- _ -- 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 -- Kristian Kielhofner -- _ -- 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] is g729 codec free? or under license???
When calling between two g729 client endpoints you do not need any licenses as long as no audio prompts or voicemail is evolved. Also as a sip trunk provider we offer g729 as a source and destination codec this allows you to make calls in and out using g729 (most carrier grade providers offer this option) You really only need to buy the number of g729 licenses that you will need for callers that require simultaneous transcoding. This is when a callers stream in or out will need to be converted to another codec format. This occurs when callers are jumping from say g729 to g711 or g729 to g722, g729 to gsm. If you plan things right and make sure any audio prompts your system is using are recorded in g729 as well as g711 and g722 you will reduce the number of g729 license considerable. Process that use a lot of g729 transcodes. ConfBridge uses g722 so all g729 has to be converted to and from g722 so 10 g729 callers to a confbridge would likely require 10 codecs (**See confbridge trick below). If you have prompts that are not pre-encoded in g729 those would use a transcoder license while playing. Voicemail would require a license as g729 has to be transcoded to one of the storage formats. The real number is based on how you are using your system. ConfBridge Trick - Have seen this used for voicemail as well, Make sure you test when using this method. If you can live with using higher bandwidth to the asterisk switch when using confbridges (endpoints also have to support in call reinvites correctly) you can force endpoints to re-invite to g722 before dropping into the conference bridge. This has the upside of not needing to transcode on the server thus improving performance and reducing g729 license requirements. This comes at the cost of needing higher bandwidth between the client endpoints and the phone. Figure about double the bandwidth when using this method. It may or may not be worth it to you depending on your scenario. Please let us know if this information helps you. Thanks Bryant Zimmerman Sr. Systems Architect Grand Dial Communications , A ZK Tech Inc. Company 616-299-5607 (mobile) 616-855-1030 Ext. 2003 (office) From: "Don Kelly" Sent: Wednesday, October 2, 2013 9:30 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] is g729 codec free? or under license??? In your scenario, all the calls are from endpoints on 181 to endpoints on 183. If the endpoint devices are similar, it seems to me that there should be no need to transcode-you can use a codec common to the endpoints. 729 would not be required. --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of s m Sent: Wednesday, October 02, 2013 2:34 AM To: Dominik George Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] is g729 codec free? or under license??? thank you Dominik you help me a lot. and the last question is how many license key should i buy? i read that license for g729 is per-channel but i don't understand what channel exactly means here. this is my scenario : 10endpointspbx181...pbx182...pbx183...10endpoints pbx181 and pbx183 has 10 endpoints connected to them. the call between these endpoints are established by pbx182. if i want to buy a license for pbx182, how many license key do i need? just one because i have just one connection on it? or two, because two trunks is defined on it? or as many as endpoints which are connected to each other via pbx182? please help me to clarify channel concept in my mind. thanks in advance SAM On Tue, Oct 1, 2013 at 11:34 AM, Dominik George wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Hi, >about g729, you mean if it get free g729 and all my systems (PBXs and >routers) use g729 codec for setting a call, call is set without any >problem? Yes, if all systems use g729 directly, you are ready to go. - -nik -BEGIN PGP SIGNATURE- Version: APG v1.0.8-fdroid iQFMBAEBCgA3BQJSSoHxMBxEb21pbmlrIEdlb3JnZSAobW9iaWxlIGtleSkgPG5p a0BuYXR1cmFsbmV0LmRlPgAKCRAvLbGk0zMOJYmRB/USyTbAqhAsnFZSGGjIcLK7 uQ3nsVNGcmE18LaBN/XFicwp5UjVB5Euju+fjKu1FhqAzECsAPMup/1JUytikmYz +32wV5YL1SNKMA/ddi/zvVa9qIbKA9yP1HuBilpD+W0DO3hdnzr2xrdR1S2z5PGZ pnYWsVlXbWYEslOuK1oaMqINoxWbsQulwQi86GPTCwPtZmhcLrvBm1sDFxWb/oPP lsPy33ZH5BeQ/XEf6nWfoiEu4Hk2S0brCH74zsz9uD6PKL1CFdLcpWv/4k5M+Mly At2PC+leZZ/TX3VNqbasslQkyv/QLZIQVtG0qQ7DGflnkrzNi5/pNV7CVT5sdPQ= =5rja -END PGP SIGNATURE- -- _ -- 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] Mail list settings?
Hey all For some reason the mailing list is sending all messages from the sending party. This makes it less than ideal when responding; as selecting reply goes to the person and not the list. Can we have it set back to the old way please? Thanks Andrew for pointing this out to me. Bryant -- _ -- 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
