Re: [asterisk-users] Timer1 RFC and SIP.CONF
6 jul 2012 kl. 09:29 skrev Elliot Murdock: Hello, Thank you for the clarification. Just a few questions: 1. What is the Timer1 used for? Timer1 is the base for many other SIP timers and it's an estimate of the roundtrip time for a packet between two SIP devices or servers. TimerB is based on T1, like the retransmit timers. 2. Since timerb is for all responses, final and provisional, the comment in sip.conf perhaps should point that out instead of implying only for provisional responses: If a provisional response is not received in this amount of time, the call will autocongest Yes, that should propably change. /O Thanks, Elliot On Thu, Jul 5, 2012 at 12:31 PM, Olle E. Johansson o...@edvina.net wrote: 4 jul 2012 kl. 13:32 skrev Elliot Murdock: Hello, I am trying to get clarity with the sip.conf timer configuration. The current configuration states: ;--- SIP timers ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;timert1=500; Default T1 timer ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 However, according to RFC 3261: (EXCERPT 17.1.1.1) T1 is an estimate of the round-trip time (RTT), and it defaults to 500 ms. Nearly all of the transaction timers described here scale with T1, and changing T1 adjusts their values. The request is not retransmitted over reliable transports. After receiving a 1xx response, any retransmissions cease altogether, and the client waits for further responses. The server transaction can send additional 1xx responses, which are not transmitted reliably by the server transaction. Eventually, the server transaction decides to send a final response. (EXCERPT 13.2.2.4 2xx Responses) The UAC core considers the INVITE transaction completed 64*T1 seconds after the reception of the first 2xx response. According to the RFC, the 64*t1 timeout is not for provisional responses, but for final responses. This seems to be in contradiction to what is stated in the sip.conf file. Unless you have PRACK support, which Asterisk not yet has, there's no timeout in the SIP protocol for provisional responses more than the need to update your provisional response at least every minute not to hit a 3 minute timeout in the SIP transaction state in a proxy. Now, the timerb is used to get ANY response from a server out there, including provisional responses. If we don't get ANY response within TIMERB, the SIP transaction dies and in a UA with a transaction layer we generate a local 408 timeout. In Asterisk, we congest the call. So the 64*T1 is for any response, including final response. It's there to decide whether or not you have intelligent SIP life forms handling your SIP request in the network universe. I hope this clears up your confusion. Regards, /Olle -- _ -- 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 --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf and binaddr issue
6 jul 2012 kl. 23:18 skrev Felix Salfelder: Hi there. i am seriously stuck with an asterisk and sip problem. the following sip.conf works with respect to some_peer: [general] bindaddr = x.y.z.w nat = no [some_peer] type=peer host=somehost secret=somesecret some other unrelated options here x.y.z.w is the ip address of the interface pointing to the network containing somehost. more precisely its the address of tun0 and route -n prints Destination Gateway Genmask Flags Metric RefUse Iface [..] x.y.z.0 0.0.0.0 255.255.255.0 U 0 0 0 tun0 [..] here 'it works' implies that i have to change and reload sip.conf after ifup tun0, or anything that forces tun0 to go down, like my dsl provider. also, the bindaddr line is suboptimal for the other peers... the same thing -- without the bindaddr part -- doesnt work. more precisely it almost works. its just incoming sound that doesnt. this must have something to do with how asterisk picks up interface addresses and communicates them to the peer in question. inspecting the packages sent to somehost, gave me the impression that asterisk uses the ip adress of ppp0 (a dsl modem) instead. how am i supposed to tell asterisk to use tun0 as the interface for [some_peer] so i can remove the bindaddr line? i've found many nat-related options in the manual, but there is no nat involved here. also, i couldnt find anything similar to iface=tun0, although the sip dialogue apparently relies on ip adresses and routing. this is about asterisk on sid, version 1:1.8.13.0~dfsg-1, but of course i'm going to switch to whatever you might suggest. The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. In fact, it's a well known bug that if you have multiple interfaces with different IP networks, Asterisk will send from the wrong IP on some of the interfaces. Sorry, /O -- _ -- 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] Rookie / sip and extensions
7 jul 2012 kl. 21:07 skrev Mikhail Lischuk: Thomas Perron писал 07.07.2012 21:48: exten = s,n,Dial(SIP/16175551212) sip.conf [general] ;register = 125010155:funnyti...@sip3.voipvoip.com/125010155 register = 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11 ; [incoming] username=125010155 I dont know what you are trying to do, but: 1) Peer doesn't have to be the same name as context. Change [incoming] in sip.conf to something like [voipvip] - it will be easier later when you have more peers. 2) What is 16175551212 ? You don't have such peer in sip.conf. If it's a number, Dial should be SIP/peer/number, for example SIP/voipvip/617 or whatever you want to dial 3) If you've posted your real password here - I strongly suggest you change it right now Please note that the account name is the name within square brackets. The username= option (now renamed to defaultuser= ) is a very different thing, and NOT the username of the account. /O -- _ -- 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] seems like call is picked and returned to me
9 jul 2012 kl. 15:24 skrev Sergio Serrano: Hi all I hope that someone of you can solve this. Right now I'm stuck! I'm using asterisk with some SIP extensions. Basically I want to establish a call between desktop voip phone (ext 181) and embedded sip system (ext 182) All I can see in CLI is: == Using SIP RTP CoS mark 5 -- Executing [182@default:1] Dial(SIP/181-000a, SIP/182) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/182 -- SIP/182-000b is ringing -- SIP/182-000b is making progress passing it to SIP/181-000a -- SIP/182-000b answered SIP/181-000a -- Remotely bridging SIP/181-000a and SIP/182-000b == Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-000a' Seems like extension 182 (called ext) is getting call and passing them another time to me 181 (origin call) I've try it with siemens pbx and works as expected It's very hard to see what's happening without seeing the SIP logs. You just see that something went wrong in the process of setting up the bridge. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf and bindaddr issue
On Tue, Jul 10, 2012 at 10:24:18AM +0200, Olle E. Johansson wrote: The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. Thanks for the reply. in the meantime i've found a sort of workaround. [general] host = dynamic ; take some local, static address bindaddr = 192.168.1.1 ; and don't use that address very much localnet = 192.168.0.0/255.255.0.0 ; ... [sip_out] ; pretend nat nat = route ; ... i'm not sure about all implications. for example, incoming connections must be handled with iptables, and in the first second of a call (from sip_out) theres no sound. i can live with that for a while. In fact, it's a well known bug that if you have multiple interfaces with different IP networks, Asterisk will send from the wrong IP on some of the interfaces. couldn't find much about it on the net. anyway, if it's well known: what would be the downside of just (silently, implicitly) taking the right adress? like when when resolving the peer-host, take a look into the routing table...? regards felix -- _ -- 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] NO AUDIO
Hi, I have a server running at more than two years with Asterisk 1.6, and began presenting problem seedlings links in external SIP extensions on some links. By doing rtp set debug on discovered the problem, he is trying to deliver the audio directly to internal IP Extension. And sometimes shown correctly on the external IP, where this time the link works correctly. These extensions are at Stake sip configured with nat = yes and = no canreivinte. In the sip show settings I have my ip correctly list externip: MY_IP: 5060 Server is not behind NAT, the IP is directly on it, it has only one network card, the SIM remote extensions are behind NAT. What may be occurring in some links for it to work correctly and not others? -- Atenciosamente Daviramos Roussenq Fortunato -- _ -- 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] channel not available and jump to next group channels
Hi list,
TRUNkA=Dahdi/g0 {g0=1-15,17-31}
TRUNKB=Dahdi/g1 {g1=32-46,48-62}
I have 2 gsm channel banks its E1 connection , its connected to server. I
define this 2 different trunks.
for example like TrunkA,TrunkB.
TRUNKA connected 1 gsm channel bank and TRUNKB connected 2 gsm channel
bank. if TRUNKA channels are not available its needs to automatically
TRUNKB. How its possible to do with Dialplan without macros.
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Re: [asterisk-users] NO AUDIO
On Tue, Jul 10, 2012 at 10:45 AM, Daviramos Roussenq Fortunato daviramo...@gmail.com wrote: I have a server running at more than two years with Asterisk 1.6, and began presenting problem seedlings links in external SIP extensions on some links. By doing rtp set debug on discovered the problem, he is trying to deliver the audio directly to internal IP Extension. And sometimes shown correctly on the external IP, where this time the link works correctly. These extensions are at Stake sip configured with nat = yes and = no canreivinte. In the sip show settings I have my ip correctly list externip: MY_IP: 5060 Server is not behind NAT, the IP is directly on it, it has only one network card, the SIM remote extensions are behind NAT. What may be occurring in some links for it to work correctly and not others? I don't know if it will resolve, but try add the localnet option in sip.conf. -- thiagoc O povo não deveria temer o governo. O governo é quem deveria temer o povo. V de Vingança -- _ -- 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] channel not available and jump to next group channels
Channels can be in more than one group.
Make g0=1-15,17-31,32-46,48-62 and
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta
Sent: Tuesday, July 10, 2012 10:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] channel not available and jump to next group channels
Hi list,
TRUNkA=Dahdi/g0 {g0=1-15,17-31}
TRUNKB=Dahdi/g1 {g1=32-46,48-62}
I have 2 gsm channel banks its E1 connection , its connected to server. I
define this 2 different trunks.
for example like TrunkA,TrunkB.
TRUNKA connected 1 gsm channel bank and TRUNKB connected 2 gsm channel bank. if
TRUNKA channels are not available its needs to automatically TRUNKB. How its
possible to do with Dialplan without macros.
--
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Re: [asterisk-users] channel not available and jump to next group channels
Thank you for your reply.
not like this. because there is 2 different types of calling is there thats
why .
On Tue, Jul 10, 2012 at 7:39 PM, Eric Wieling ewiel...@nyigc.com wrote:
Channels can be in more than one group.
Make g0=1-15,17-31,32-46,48-62 and
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta
Sent: Tuesday, July 10, 2012 10:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] channel not available and jump to next group
channels
Hi list,
TRUNkA=Dahdi/g0 {g0=1-15,17-31}
TRUNKB=Dahdi/g1 {g1=32-46,48-62}
I have 2 gsm channel banks its E1 connection , its connected to server. I
define this 2 different trunks.
for example like TrunkA,TrunkB.
TRUNKA connected 1 gsm channel bank and TRUNKB connected 2 gsm channel
bank. if TRUNKA channels are not available its needs to automatically
TRUNKB. How its possible to do with Dialplan without macros.
--
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[asterisk-users] connections to manager
Is there a limit to the number of connections that manager can handle at one time? In my logs I see connect error but then try again in a few seconds and it works. I could have quite a number of connections at one time. How can I up the limit. Jerry -- _ -- 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] connections to manager
Is there a limit to the number of connections that manager can handle at one time? In my logs I see connect error but then try again in a few seconds and it works. I could have quite a number of connections at one time. How can I up the limit. Jerry -- _ -- 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] 10.6.0-rc2: tmp full of core.PBX
I've installed 10.6.0-rc2 on two machines. On one of the machines (but not the other) /tmp gets filled with: ... -rw---. 1 asterisk asterisk 53661696 Jul 7 23:46 core.PBX-2012-07-07T23:46:10-0400 -rw---. 1 asterisk asterisk 53891072 Jul 7 23:48 core.PBX-2012-07-07T23:48:55-0400 -rw---. 1 asterisk asterisk 53469184 Jul 7 23:53 core.PBX-2012-07-07T23:53:00-0400 -rw---. 1 asterisk asterisk 53739520 Jul 7 23:58 core.PBX-2012-07-07T23:58:25-0400 ... and finally fills up all the space. grep -v ';' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error messages = notice,warning,error Any clue what to look for? sean -- _ -- 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] 10.6.0-rc2: tmp full of core.PBX
- Original Message - From: sean darcy seandar...@gmail.com To: asterisk-users@lists.digium.com Sent: Tuesday, July 10, 2012 10:42:20 AM Subject: [asterisk-users] 10.6.0-rc2: tmp full of core.PBX I've installed 10.6.0-rc2 on two machines. On one of the machines (but not the other) /tmp gets filled with: ... -rw---. 1 asterisk asterisk 53661696 Jul 7 23:46 core.PBX-2012-07-07T23:46:10-0400 -rw---. 1 asterisk asterisk 53891072 Jul 7 23:48 core.PBX-2012-07-07T23:48:55-0400 -rw---. 1 asterisk asterisk 53469184 Jul 7 23:53 core.PBX-2012-07-07T23:53:00-0400 -rw---. 1 asterisk asterisk 53739520 Jul 7 23:58 core.PBX-2012-07-07T23:58:25-0400 ... and finally fills up all the space. grep -v ';' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error messages = notice,warning,error Any clue what to look for? sean You'll need to provide a backtrace using the instructions below: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace As soon as you have the information, please open an issue in JIRA. Thanks! -- Matthew Jordan Digium, Inc. | Software Developer 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
Re: [asterisk-users] 10.6.0-rc2: tmp full of core.PBX
On 07/10/2012 11:44 AM, Matthew Jordan wrote: - Original Message - From: sean darcy seandar...@gmail.com To: asterisk-users@lists.digium.com Sent: Tuesday, July 10, 2012 10:42:20 AM Subject: [asterisk-users] 10.6.0-rc2: tmp full of core.PBX I've installed 10.6.0-rc2 on two machines. On one of the machines (but not the other) /tmp gets filled with: ... -rw---. 1 asterisk asterisk 53661696 Jul 7 23:46 core.PBX-2012-07-07T23:46:10-0400 -rw---. 1 asterisk asterisk 53891072 Jul 7 23:48 core.PBX-2012-07-07T23:48:55-0400 -rw---. 1 asterisk asterisk 53469184 Jul 7 23:53 core.PBX-2012-07-07T23:53:00-0400 -rw---. 1 asterisk asterisk 53739520 Jul 7 23:58 core.PBX-2012-07-07T23:58:25-0400 ... and finally fills up all the space. grep -v ';' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error messages = notice,warning,error Any clue what to look for? sean You'll need to provide a backtrace using the instructions below: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace As soon as you have the information, please open an issue in JIRA. Thanks! -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org Nothing in /tmp since I erased all the core* files this morning. If it starts again, I'll do a backtrace. sean -- _ -- 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] Flowroute: howto set outbound callerid (ast 1.4)?
Hi, The flowroute website mentions that they set callerid on outbound calls based on the presence of (in order of preference): P-Asserted-Identity, Remote-Party-ID or From:. I've been trying to make outbound callerid work via flowroute to no avail. Does anyone have an extensions.conf / sip.conf snippet howto make this work? This is for Asterisk 1.4.44. Thanks! Patrick -- _ -- 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] channel not available and jump to next group channels
On Tue, Jul 10, 2012 at 9:04 AM, mahesh katta maheshka...@flexydial.comwrote:
Hi list,
TRUNkA=Dahdi/g0 {g0=1-15,17-31}
TRUNKB=Dahdi/g1 {g1=32-46,48-62}
I have 2 gsm channel banks its E1 connection , its connected to server. I
define this 2 different trunks.
for example like TrunkA,TrunkB.
TRUNKA connected 1 gsm channel bank and TRUNKB connected 2 gsm channel
bank. if TRUNKA channels are not available its needs to automatically
TRUNKB. How its possible to do with Dialplan without macros.
exten = _XX,n,Dial(${TRUNKA}/${EXTEN})
exten = _XX,n,Dial(${TRUNKB}/${EXTEN})
Swap _XX for whatever your outbound extensions would be...
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?
SIPAddHeader() comes to mind. :-) -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:Hi, The flowroute website mentions that they set callerid on outbound calls based on the presence of (in order of preference): P-Asserted-Identity, Remote-Party-ID or From:. I've been trying to make outbound callerid work via flowroute to no avail. Does anyone have an extensions.conf / sip.conf snippet howto make this work? This is for Asterisk 1.4.44. Thanks! Patrick -- _ -- 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] Flowroute: howto set outbound callerid (ast 1.4)?
On 10-07-12 18:29, Alex Balashov wrote: SIPAddHeader() comes to mind. :-) Yup I got that far :) I tried things like (with correct name number): exten = _1ZX,1,SipAddHeader(P-Asserted-Identity: Global Minties Corp sip:19995551212@AST_BOX_FQDN) But that did not work as flowroute always sends VOIP CALLER and a ton of different numbers on outbound calls. So I guess I am doing something wrong but I can't figure out what. Regards, Patrick -- _ -- 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] Flowroute: howto set outbound callerid (ast 1.4)?
At 09:20 AM 7/10/2012, you wrote: I've been trying to make outbound callerid work via flowroute to no avail. Does anyone have an extensions.conf / sip.conf snippet howto make this work? This is for Asterisk 1.4.44. This is a section of code I use to choose outgoing callerid for my Flowroute lines. I have a number of companies and this lets the caller select what the called parts sees. Ira same = n(got0),set(thiscid=NOONE2345678901) same = n,goto(gotcallerid) same = n(got1),set(thiscid=Bob and Lucy3124726322) same = n,goto(gotcallerid) same = n(got2),set(thiscid=31247240223124724022) same = n,goto(gotcallerid) same = n(got3),set(thiscid=Mustang3126925021) same = n,goto(gotcallerid) -- _ -- 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] Flowroute: howto set outbound callerid (ast 1.4)?
Check your users.conf - this looks like an override issue to me. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists Sent: Tuesday, July 10, 2012 11:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)? On 10-07-12 18:29, Alex Balashov wrote: SIPAddHeader() comes to mind. :-) Yup I got that far :) I tried things like (with correct name number): exten = _1ZX,1,SipAddHeader(P-Asserted-Identity: Global Minties Corp sip:19995551212@AST_BOX_FQDN) But that did not work as flowroute always sends VOIP CALLER and a ton of different numbers on outbound calls. So I guess I am doing something wrong but I can't figure out what. Regards, Patrick -- _ -- 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] Flowroute: howto set outbound callerid (ast 1.4)?
On Tue, Jul 10, 2012 at 11:45 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 10-07-12 18:29, Alex Balashov wrote: SIPAddHeader() comes to mind. :-) Yup I got that far :) I tried things like (with correct name number): exten = _1ZX,1,SipAddHeader(P-**Asserted-Identity: Global Minties Corp sip:19995551212@AST_BOX_FQDN**) But that did not work as flowroute always sends VOIP CALLER and a ton of different numbers on outbound calls. So I guess I am doing something wrong but I can't figure out what. You can't* set the outbound name. That's defined in the national caller id name database that the receiving phone company dips into. As far as I know, Flowroute does not add entries to this database, nor do they dip it when you receive a call to pass the caller ID name on inbound calls. Other providers do. * - You may be able to set it if you're calling other users on the Flowroute network, I'm not sure. But in general, once your call leaves the Flowroute network, the only way to get the CNAM info is from a CNAM dip to the national database (I don't recall the actual name). -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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
Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?
On 10-07-12 18:49, Warren Selby wrote: You can't* set the outbound name. That's defined in the national caller id name database that the receiving phone company dips into. As far as I know, Flowroute does not add entries to this database, nor do they dip it when you receive a call to pass the caller ID name on inbound calls. Other providers do. Thank you for your feedback Warren. I removed the outbound name but still get random numbers VOIP CALLER on outbound calls. Googling I tried some more: SipAddHeader(P-Asserted-Identity: sip:19995551212@AST_BOX_FQDN) SipAddHeader(P-Asserted-Identity: 19995551212) SipAddHeader(P-Preferred-Identity: sip:19995551212@AST_BOX_FQDN) SipAddHeader(P-Preferred-Identity: 19995551212) But none of them work. So unless someone has the magic incantation howto make this work I'll open a ticket with flowroute. Regards, Patrick -- _ -- 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] Trixbox or FreePBX or Elastix or PBX In a Flash
- Original Message - Thanks Tim. One of the problem that I am facing is the complicated generated configuration for the FreePBX, is it the same thing in the Elastix? To understand this complicated generated commands, is there a documentation to explain this for FreePBX or Elastix? One of my friend told me that he installed (as I remember) FreePBX and there were already existed the TFTP files that the Cisco IP Phones is requesting (for sip or skiny) and already there were a TFTP server. Which module to do this? FreePBX (and any other project based on it, including Trixbox, Elastix, PIAF, AsteriskNow, etc) stores all information in a MySQL database, then when you click 'apply changes' it takes all of the system config info from the database, and generates the Asterisk dialplan code in /etc/asterisk . This is the tradeoff for having a magical GUI do most of the work, and doing everything by hand. You can of course make your own changes that will not be overwritten by editing the appropriate *_custom.conf files. For example, to add contexts and/or dialplan stuff, put it in extensions_custom.conf. The same applies for sip_custom.conf, etc... Most of the predone projects (Elastix is my favorite at the moment) include some sort of endpoint manager that will generate configs for your phones. I'm not sure specifically on Cisco phones, other than they are a huge PITA in general. The system just needs a TFTP server installed, and the phones pointed to it (manually, or by using DHCP option 66). --Tim -- _ -- 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] Flowroute: howto set outbound callerid (ast 1.4)?
On Tue, Jul 10, 2012 at 12:34 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
Thank you for your feedback Warren. I removed the outbound name but still
get random numbers VOIP CALLER on outbound calls. Googling I tried some
more:
SipAddHeader(P-Asserted-**Identity: sip:19995551212@AST_BOX_FQDN**)
SipAddHeader(P-Asserted-**Identity: 19995551212)
SipAddHeader(P-Preferred-**Identity: sip:19995551212@AST_BOX_FQDN**)
SipAddHeader(P-Preferred-**Identity: 19995551212)
But none of them work. So unless someone has the magic incantation howto
make this work I'll open a ticket with flowroute.
I use Flowroute. My outbound callerID is set as follows:
[outgoing]
exten = _X.,1,Verbose(Outound call from ${callidnum} to ${EXTEN} on
${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)})
exten = _X.,n,Set(CALLERID(num)=${callidnum})
exten = _X.,n,Goto(outgoing-dial,${EXTEN},1)
[outgoing-dial]
exten = _NXXNXX,1,Dial(SIP/1${EXTEN}@flowroute)
exten = _1NXXNXX,1,Dial(SIP/${EXTEN}@flowroute)
${callidnum} is a variable from my SIP peer (setvar=callidnum=7133437300).
This always passes my proper phone number when I make outbound calls.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash
On Tue, Jul 10, 2012 at 12:39 PM, Tim Nelson tnel...@rockbochs.com wrote: Most of the predone projects (Elastix is my favorite at the moment) include some sort of endpoint manager that will generate configs for your phones. I'm not sure specifically on Cisco phones, other than they are a huge PITA in general. The system just needs a TFTP server installed, and the phones pointed to it (manually, or by using DHCP option 66). Speaking from experience - I have a client that has Trixbox (setup by a previous phone consultant) that also uses all 7960 and 7940 phones. Trixbox works perfectly with the phones - it uses an endpoint manager that automatically discovers and configures the phones, even setting the background display. I think it's a special add-on module that was added to this installation, I'm not entirely sure. I can check if you would like me to pursue this? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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
Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?
On 10-07-12 18:48, Danny Nicholas wrote: Check your users.conf - this looks like an override issue to me. Thank you for your feedback Danny. users.conf is default and has not been touched. Regards, Patrick -- _ -- 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] Flowroute: howto set outbound callerid (ast 1.4)?
On 10-07-12 18:47, Ira wrote: At 09:20 AM 7/10/2012, you wrote: I've been trying to make outbound callerid work via flowroute to no avail. Does anyone have an extensions.conf / sip.conf snippet howto make this work? This is for Asterisk 1.4.44. This is a section of code I use to choose outgoing callerid for my Flowroute lines. I have a number of companies and this lets the caller select what the called parts sees. Ira same = n(got0),set(thiscid=NOONE2345678901) same = n,goto(gotcallerid) same = n(got1),set(thiscid=Bob and Lucy3124726322) same = n,goto(gotcallerid) same = n(got2),set(thiscid=31247240223124724022) same = n,goto(gotcallerid) same = n(got3),set(thiscid=Mustang3126925021) same = n,goto(gotcallerid) Thank you for that snippet Ira. You and Warren were spot on. All is fine now. Thanks! Patrick -- _ -- 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] Flowroute: howto set outbound callerid (ast 1.4)?
On 10-07-12 19:48, Warren Selby wrote:
On Tue, Jul 10, 2012 at 12:34 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl
mailto:asterisk-l...@puzzled.xs4all.nl wrote:
Thank you for your feedback Warren. I removed the outbound name but
still get random numbers VOIP CALLER on outbound calls. Googling
I tried some more:
SipAddHeader(P-Asserted-__Identity: sip:19995551212@AST_BOX_FQDN__)
SipAddHeader(P-Asserted-__Identity: 19995551212)
SipAddHeader(P-Preferred-__Identity: sip:19995551212@AST_BOX_FQDN__)
SipAddHeader(P-Preferred-__Identity: 19995551212)
But none of them work. So unless someone has the magic incantation
howto make this work I'll open a ticket with flowroute.
I use Flowroute. My outbound callerID is set as follows:
[outgoing]
exten = _X.,1,Verbose(Outound call from ${callidnum} to ${EXTEN} on
${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)})
exten = _X.,n,Set(CALLERID(num)=${callidnum})
exten = _X.,n,Goto(outgoing-dial,${EXTEN},1)
[outgoing-dial]
exten = _NXXNXX,1,Dial(SIP/1${EXTEN}@flowroute)
exten = _1NXXNXX,1,Dial(SIP/${EXTEN}@flowroute)
${callidnum} is a variable from my SIP peer
(setvar=callidnum=7133437300). This always passes my proper phone
number when I make outbound calls.
Thank you for that snippet Warren. I setup a different US DID and called
that one via flowroute and the callerid worked. Previously I called a
voip.ms toll-free number. I'll blame it on (lack of) carrier
interoperability :) Good to know outbound callerid works without having
to use magic SipAddHeader incantations.
Thanks!
Patrick
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Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash
I'm currently trying to decide on which GUI-enabled version of Asterisk to use for one particular installation, where we will need good telecommuter support. We've made it so easy for people to work remotely that the customer is downsizing their real estate and will have 90% remote workers with them rotating through the office as needed. So most phones in the office will be shared, and I'm looking for a version of Asterisk that will easily allow people to log in and out of a specific desk. What are your suggestions? I have very little experience with GUI versions of Asterisk; we use bare Asterisk for nearly everything. -- _ -- 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] Trixbox or FreePBX or Elastix or PBX In a Flash
On 07/10/2012 01:42 PM, Carlos Alvarez wrote: I'm currently trying to decide on which GUI-enabled version of Asterisk to use for one particular installation, where we will need good telecommuter support. We've made it so easy for people to work remotely that the customer is downsizing their real estate and will have 90% remote workers with them rotating through the office as needed. So most phones in the office will be shared, and I'm looking for a version of Asterisk that will easily allow people to log in and out of a specific desk. What are your suggestions? I have very little experience with GUI versions of Asterisk; we use bare Asterisk for nearly everything. This can be done using Digium phones; they have built-in support for selecting which 'user' they should be when they are reconfigured. It's slightly more complicated than a simple login/logout because it requires rebooting the phone, but it's there and it works. They even support the case where a user has a phone at their house, but comes into the office and 'steals' their extension away from the house phone; the house phone will then go into an automatic reconfiguration state and wait for someone tell it which extension it should 'be'... when the user returns home, they can 'steal' the extension back from the office. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash
- Original Message - I'm currently trying to decide on which GUI-enabled version of Asterisk to use for one particular installation, where we will need good telecommuter support. We've made it so easy for people to work remotely that the customer is downsizing their real estate and will have 90% remote workers with them rotating through the office as needed. So most phones in the office will be shared, and I'm looking for a version of Asterisk that will easily allow people to log in and out of a specific desk. What are your suggestions? I have very little experience with GUI versions of Asterisk; we use bare Asterisk for nearly everything. Not to sound like a broken record or anything... but I'd say give Elastix a go. It is top notch in terms of release quality and features. And, being based on FreePBX, you can set it to 'Device and User' mode instead of the default extensions mode so users can 'hotdesk' between phones. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf and binaddr issue
On 07/10/2012 03:24 AM, Olle E. Johansson wrote: The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. In fact, it's a well known bug that if you have multiple interfaces with different IP networks, Asterisk will send from the wrong IP on some of the interfaces. Are you sure about that? The only problem area that I'm aware of is when there are multiple *overlapping* interfaces (on the same subnet, or providing the same route(s)). In that case, Asterisk can receive messages on one IP address out of the overlapping set, but reply using a different one from the set, because it doesn't specify the source IP address and instead lets the UDP/IP stack select one. If the interfaces don't overlap in any way, I don't see how it would be possible for Asterisk to send messages with the wrong source IP address, since it does not specify the source IP address at all. If this is occurring, it must involve the operating system's IP stack in some fashion. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash
On Tue, Jul 10, 2012 at 11:46 AM, Kevin P. Fleming kpflem...@digium.comwrote: This can be done using Digium phones; they have built-in support for selecting which 'user' they should be when they are reconfigured. It's slightly more complicated than a simple login/logout because it requires rebooting the phone, but it's there and it works. They even support the case where a user has a phone at their house, but comes into the office and 'steals' their extension away from the house phone; the house phone will then go into an automatic reconfiguration state and wait for someone tell it which extension it should 'be'... when the user returns home, they can 'steal' the extension back from the office. Unfortunately, telling them to dump 50-some Polycom phones that still work is a tough one. Not impossible though. So if I understand correctly, the user reboots the phone when he arrives, and goes into the configure mode? Is it end user friendly, or complex? I still don't have a server with the Digium phone module on it to test. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Trixbox or FreePBX or Elastix or PBX In a Flash
On 10-07-12 20:42, Carlos Alvarez wrote: I'm currently trying to decide on which GUI-enabled version of Asterisk to use for one particular installation, where we will need good telecommuter support. We've made it so easy for people to work remotely that the customer is downsizing their real estate and will have 90% remote workers with them rotating through the office as needed. So most phones in the office will be shared, and I'm looking for a version of Asterisk that will easily allow people to log in and out of a specific desk. What are your suggestions? I have very little experience with GUI versions of Asterisk; we use bare Asterisk for nearly everything. Afaik development of the free Trixbox has stalled. Elastix is a popular alternative and iirc it comes with provisioning tools that work with Polycom. Or you could just install a bare Asterisk and slap FreePBX on top of it if you don't need all the fancy stuff that Elastix offers. Regards, Patrick -- _ -- 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] Trixbox or FreePBX or Elastix or PBX In a Flash
On 07/10/2012 01:50 PM, Carlos Alvarez wrote: On Tue, Jul 10, 2012 at 11:46 AM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: This can be done using Digium phones; they have built-in support for selecting which 'user' they should be when they are reconfigured. It's slightly more complicated than a simple login/logout because it requires rebooting the phone, but it's there and it works. They even support the case where a user has a phone at their house, but comes into the office and 'steals' their extension away from the house phone; the house phone will then go into an automatic reconfiguration state and wait for someone tell it which extension it should 'be'... when the user returns home, they can 'steal' the extension back from the office. Unfortunately, telling them to dump 50-some Polycom phones that still work is a tough one. Not impossible though. So if I understand correctly, the user reboots the phone when he arrives, and goes into the configure mode? Is it end user friendly, or complex? I still don't have a server with the Digium phone module on it to test. That's basically it, yes. I just did this on my D70: * Press Menu, select Restart, press Yes. * During the restart, while the progress bar is progressing, push a key. This results in 'Reconfiguration enabled' on the screen. * When the server list appears, choose one (if there is only one server, this will be skipped). * Choose a user from the user list and enter the PIN for it; alternatively, if a global PIN is required by the server, it will need to be entered before the user list will be displayed. * Wait for the phone to configure itself. This takes about 60 seconds. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk 1.8.14.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.14.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- format_mp3: Fix a possible crash in mp3_read(). (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk) * --- Fix local channel chains optimizing themselves out of a call. (Closes issue ASTERISK-16711. Reported by Alec Davis) * --- Update a peer's LastMsgsSent when the peer is notified of waiting messages (Closes issue ASTERISK-17866. Reported by Steve Davies) * --- Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt. (Closes issue ASTERISK-19425. Reported by David Cunningham) * --- Send more accurate identification information in dialog-info SIP NOTIFYs. (Closes issue ASTERISK-16735. Reported by Maciej Krajewski) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.14.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10.6.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.6.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- format_mp3: Fix a possible crash in mp3_read(). (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk) * --- Fix local channel chains optimizing themselves out of a call. (Closes issue ASTERISK-16711. Reported by Alec Davis) * --- Re-add LastMsgsSent value for SIP peers (Closes issue ASTERISK-17866. Reported by Steve Davies) * --- Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt. (Closes issue ASTERISK-19425. Reported by David Cunningham) * --- Send more accurate identification information in dialog-info SIP NOTIFYs. (Closes issue ASTERISK-16735. Reported by Maciej Krajewski) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash
On 10/7/12 7:46 pm, Tim Nelson wrote: Not to sound like a broken record or anything... but I'd say give Elastix a go. It is top notch in terms of release quality and features. And, being based on FreePBX, you can set it to 'Device and User' mode instead of the default extensions mode so users can 'hotdesk' between phones. We have a number of deployments using the FPBX 'Device and User' mode in a similar manner to that the OP requested, and they seem to work fairly well. On 10/7/12 7:58 pm, Patrick Lists wrote: Or you could just install a bare Asterisk and slap FreePBX on top of it Personally, I'd go down this route, especially if you're already familiar with Linux (which I'm guessing you are if you're used to working with bare Asterisk). This way you can choose the distro you like best rather than having to adapt to whatever the all-in-one maintainer has chosen to use. It also opens up options if you find you need to run other packages on the same server at any point. 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
Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash
Recent Polycom firmware versions (4.x, I think) also have support for user sort of stuff. See the 4.x Admin Guide. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, July 10, 2012 3:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash On 07/10/2012 01:50 PM, Carlos Alvarez wrote: On Tue, Jul 10, 2012 at 11:46 AM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: This can be done using Digium phones; they have built-in support for selecting which 'user' they should be when they are reconfigured. It's slightly more complicated than a simple login/logout because it requires rebooting the phone, but it's there and it works. They even support the case where a user has a phone at their house, but comes into the office and 'steals' their extension away from the house phone; the house phone will then go into an automatic reconfiguration state and wait for someone tell it which extension it should 'be'... when the user returns home, they can 'steal' the extension back from the office. Unfortunately, telling them to dump 50-some Polycom phones that still work is a tough one. Not impossible though. So if I understand correctly, the user reboots the phone when he arrives, and goes into the configure mode? Is it end user friendly, or complex? I still don't have a server with the Digium phone module on it to test. That's basically it, yes. I just did this on my D70: * Press Menu, select Restart, press Yes. * During the restart, while the progress bar is progressing, push a key. This results in 'Reconfiguration enabled' on the screen. * When the server list appears, choose one (if there is only one server, this will be skipped). * Choose a user from the user list and enter the PIN for it; alternatively, if a global PIN is required by the server, it will need to be entered before the user list will be displayed. * Wait for the phone to configure itself. This takes about 60 seconds. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
[asterisk-users] Planned service outage for community services on July 12, 2012
On July 12, 2012 from approximately 11:00AM to 11:30AM (Central Daylight Time, GMT-5), the core routers that provide connectivity through to all Asterisk community services will be swapped out. This will mean that these services will be unavailable during most, if not all, of this time window. Once the move is complete, the services will be available again, with no user-visible changes. The services affected include: bamboo.asterisk.org code.asterisk.org downloads.digium.com downloads.asterisk.org git.asterisk.org issues.asterisk.org packages.asterisk.org reviewboard.asterisk.org svn.asterisk.org svnview.digium.com wiki.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
Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration
I went for admin/module admin and I search for custom contexts but did not find it. How I can get it? Regards Bilal --- The module is custom contexts - its a third party option in the module admin But you can write contexts in the extensions_custom.conf if you want to I wouldn't use freepbx to generate your code - its quite complex code for a roll your own system, but very useful if you learn its gui and options Also you can limit outbound routes to certain extension ranges which can avoid the need for contexts but its up to you Cheers Duncan On 6/07/2012, at 4:20 PM, SamyGo wrote: Hey, If you want to have all the dialplan features for your extensions and still need to implement some outbound calling restrictions then you need to look for some modules in freePBX. i've used that module exactly for this purpose and it works..can't remember its name. Just google it or lookup the latest modules available. Regards, Sammy On Fri, Jul 6, 2012 at 3:20 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I have to write the configuration in my hand and it will not be autogeneration, correct? In this case, the Phone will not have any of the features that I am going to add it in the GUI because these features will be in the default context which is not included (unless I add it manually) in the context that I will set it. Also, if I set the context and I write manually the configuration for this context, I do not think that I will have CDR (because to have CDR, I have to use some configuration to log in the database and becoming able to see it in the CDR). Again, if I used the default context, then it is good that all the stations to have the same context and same previlages .. so it is not a practical way. So, what is the solution for this? As I see the only benifit of the Freepbx (the GUI), is to generate the configuration that I can use it when I am writing the manual configuration (by including it and so on). In this case, I am afraid that things will become maybe more complex :) !! Any advise for this? Regards Bilal -- _ -- 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] FreePBX: using context other than the default context and the generation for the configuration
Please don't top-post. On Tue, 10 Jul 2012, bilal ghayyad wrote: -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] FreePBX: using context other than the default context and the generation for the configuration
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Tuesday, July 10, 2012 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration Please don't top-post. OK. -- _ -- 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] FreePBX: using context other than the default context and the generation for the configuration
Dear Warren;
I did not understand the example below well.
What the Verbose will do? It will write in the CDR or the database? Really this
did not understand.
Also did not understand this lineL same = n,Goto(${EXTEN},from-internal,1) How
it will work? Can u plz explain?
Regards
Bilal
umm Warren, yes including from-internal is the way of
getting all the
features,,,but in my experience the calls going out
using the dialplan
script we manually enter in our custome context don't
get inserted into the
FreePBX CDR and recording stuff !!
Okay, if you're writing custom dialplan to control outbound
calling, but
you want to utilize the FreePBX standard features, without
using custom
modules, you can do something like the following, adjusting
for your
specific situations of course:
[custom-local-only]
; local NANPA calling for area code 281
exten = _281NXX,1,Verbose(Outbound call from
local-only context)
same = n,Goto(${EXTEN},from-internal,1)
; extension-to-extension (internal) calling, assuming 2XXX
internal
extension plan
exten = _2XXX,1,Verbose(Internal extension-to-extension
call)
same = n,Goto(${EXTEN},from-internal,1)
[custom-long-distance]
; long distance NANPA calling, dial a 1 to dial anything
outside of a local
number
exten = _1NXXNXX,1,Verbose(Outbound call from local
and long-distance
context)
same = n,Goto(${EXTEN},from-internal,1)
; allow local calls also, without having to dial a 1
include = custom-local-only
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration
- Original Message -
From: bilal ghayyad bilmar...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 10, 2012 4:25:40 PM
Subject: Re: [asterisk-users] FreePBX: using context other than the default
context and the generation for the
configuration
Dear Warren;
I did not understand the example below well.
What the Verbose will do? It will write in the CDR or the database?
Really this did not understand.
https://wiki.asterisk.org/wiki/display/AST/Application_Verbose
Also did not understand this lineL same =
n,Goto(${EXTEN},from-internal,1) How it will work? Can u plz
explain?
https://wiki.asterisk.org/wiki/display/AST/Application_Goto
Regards
Bilal
Its clear that you're missing some fundamental knowledge of
Asterisk configuration and operation. While its perfectly
fine to ask the mailing list for help, its also worthwhile
to learn what you can on your own.
Before continuing to ask questions on this list, you should
consider reading the information that is freely available
on the internet.
http://ofps.oreilly.com/titles/9780596517342/
(Note that the authors of that fine book would probably
appreciate you paying them for their hard work and effort,
but its still freely available if you choose not to do so)
--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] Forcing SIP trunk matching order?
On Mon, Jul 2, 2012 at 12:13 AM, Olle E. Johansson o...@edvina.net wrote: No. This is probably because you are using phone numbers as names of devices with type=friend in sip.conf. That's generally a bad idea. The SIP channel matches an incoming call this way: 1. Take the From: user name and match with the list of type=user and type=friend 2. Take the sender's IP and port and match with the list of peers 3. Send the call to the context defined in the [general] section of sip conf In Asterisk 1.4 and hopefully 1.8 the last peer in sip.conf will match first. In 1.8 the internal strcutures was changed, but I hope that this functionality was retained. We had a dicussion about it, but I personally haven't tested the result. One needs to know the matching order, so if 1.8 doesn't behave that way, we need to fix it. The recommended way is to NOT use anything that likely will end up as a caller ID as names of devices in sip.conf. The name is whatever you have within square brackets above definitions of type=friend or type=user. The username= option is another option, not the name of the device. The quick way to solve your problems is to stop using type=friend and start using type=peer instead. Hi Ollie, You are correct, I do have callerID-type names as accounts in sip.conf. The hosts are set to dynamic. Is this a problem with type=peer? Would the deny/allow suggestion posted earlier also work with type=friend? Thanks. -- James -- _ -- 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
