Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
> > As promised, here is a follow up on my quest to get CallerID > > correctly > > presented when forwarding calls to cellphones. > > > > Here is a reminder of the issue at hand: > > > > Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension) > > which forwards to Cory (GSM handset) > > What I would like to get is to see Alice's number (not Bob's number) > > presented to Cory. > > Sometimes, I get Alice's number, sometimes, I get Bob's number (new > > findings from last sunday trials). > > And of course, if Daniel or Eric would call Bob, the CallerID number > > presented to Cory would either be Daniel's number, Eric's number or > > Bob's number depending on a root cause I'm looking after for several > > days now. > > > > > > > > To check if CallerID is filtered or controlled by Telco, I > > originated > > calls from Asterisk using hand crafted caller ids: any CallerID was > > correctly presented. > > So I originally thought the root cause I'm after is a telco > > equipment > > switching ANI and CID. > > But a close look at some last trials output makes me asking for > > opinions from this list readers. > > > > Here follows, the anonymized (and hand indented) output of command > > PRI > > debug command. > > I focused on the end of call setup dialog. > > > > For the successfully presented call, the output is: > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > [6c 0b 21 83 37 38 > > 36 > > XX XX XX XX XX XX] > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Calling Number > > (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony > > Numbering Plan (E.164/E.163) (1) > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Presentation: > > Presentation allowed of network provided number (3) '78649' ] > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > [70 0b 80 30 36 37 > > 31 > > XX XX XX XX XX XX] > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Called Number > > (len=13) > > [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) > > '067100' ] > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > [74 0e 21 01 8f 33 > > 33 > > 33 34 34 XX XX XX XX XX XX] > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Redirecting Number > > (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony > > Numbering Plan (E.164/E.163) (1) > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Ext: 0 Presentation: > > Presentation permitted, user number passed network screening (1) > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Ext: 1 Reason: > > Forwarded unconditionally (15) > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ] > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > [a1] > > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Sending Complete > > (len= > > 1) > > > > > > For the unsuccessfully presented call, the output is: > > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > [6c 0b 21 83 36 37 > > 38 > > XX XX XX XX XX XX] > > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > Calling Number > > (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony > > Numbering Plan (E.164/E.163) (1) > > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > Presentation: > > Presentation allowed of network provided number (3) '67854' ] > > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > [70 0b 80 30 36 37 > > 31 > > XX XX XX XX XX XX] > > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > Called Number > > (len=13) > > [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) > > '067100' ] > > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > [a1] > > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > Sending Complete > > (len= > > 1) > > > > > > Am I correctly interpreting when saying that in the successful call, > > Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX > > XX] message which is not otherwise sent ? > > What can explains this difference ? > > Is this something I can (should) control ? > > > > For reference: > > dahdi show version > > DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC > > pri show version > > libpri version: 1.4.10.2 > > Improved support for manipulation of redirecting number is available > with the REDIRECTING dialplan function in Asterisk v1.8.x and > libpr
Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
> As promised, here is a follow up on my quest to get CallerID correctly > presented when forwarding calls to cellphones. > > Here is a reminder of the issue at hand: > > Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension) > which forwards to Cory (GSM handset) > What I would like to get is to see Alice's number (not Bob's number) > presented to Cory. > Sometimes, I get Alice's number, sometimes, I get Bob's number (new > findings from last sunday trials). > And of course, if Daniel or Eric would call Bob, the CallerID number > presented to Cory would either be Daniel's number, Eric's number or > Bob's number depending on a root cause I'm looking after for several > days now. > > > > To check if CallerID is filtered or controlled by Telco, I originated > calls from Asterisk using hand crafted caller ids: any CallerID was > correctly presented. > So I originally thought the root cause I'm after is a telco equipment > switching ANI and CID. > But a close look at some last trials output makes me asking for > opinions from this list readers. > > Here follows, the anonymized (and hand indented) output of command PRI > debug command. > I focused on the end of call setup dialog. > > For the successfully presented call, the output is: > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > [6c 0b 21 83 37 38 36 > XX XX XX XX XX XX] > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Calling Number > (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony > Numbering Plan (E.164/E.163) (1) > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Presentation: > Presentation allowed of network provided number (3) '78649' ] > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > [70 0b 80 30 36 37 31 > XX XX XX XX XX XX] > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Called Number (len=13) > [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) > '067100' ] > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > [74 0e 21 01 8f 33 33 > 33 34 34 XX XX XX XX XX XX] > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Redirecting Number > (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony > Numbering Plan (E.164/E.163) (1) > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Ext: 0 Presentation: > Presentation permitted, user number passed network screening (1) > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Ext: 1 Reason: > Forwarded unconditionally (15) > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ] > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > [a1] > [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: > Sending Complete (len= > 1) > > > For the unsuccessfully presented call, the output is: > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > [6c 0b 21 83 36 37 38 > XX XX XX XX XX XX] > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > Calling Number > (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony > Numbering Plan (E.164/E.163) (1) > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > Presentation: > Presentation allowed of network provided number (3) '67854' ] > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > [70 0b 80 30 36 37 31 > XX XX XX XX XX XX] > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > Called Number (len=13) > [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) > '067100' ] > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > [a1] > [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: > Sending Complete (len= > 1) > > > Am I correctly interpreting when saying that in the successful call, > Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX > XX] message which is not otherwise sent ? > What can explains this difference ? > Is this something I can (should) control ? > > For reference: > dahdi show version > DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC > pri show version > libpri version: 1.4.10.2 Improved support for manipulation of redirecting number is available with the REDIRECTING dialplan function in Asterisk v1.8.x and libpri v1.4.12. Prior to Asterisk v1.8.x you only have CALLERID(RDNIS). https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information Richard -- _ -- 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] Realtime Queue - changing strategy to linear needs Asterisk restart
> We have realtime queue architecture on asterisk 1.8.7.0 > I noticed that when we change strategy from any other to 'linear' it > requires Asterisk restart take the change in effect. > I have one realtime queue '1' with strategy set to 'ringall' and I > change its strategy to 'linear'. Now when check on Asterisk CLI it > shows me warning given below. > > demo*CLI> queue show 1 > 1 has 0 calls (max 500) in 'ringall' strategy (0s holdtime, 0s > talktime), W:1, C:0, A:0, SL:0.0% within 100s > No Members > No Callers > > [Nov 8 12:10:18] WARNING[4887]: app_queue.c:2034 queue_set_param: > Changing to the linear strategy currently requires asterisk to be > restarted. > [Nov 8 12:10:18] WARNING[4887]: app_queue.c:2034 queue_set_param: > Changing to the linear strategy currently requires asterisk to be > restarted. > > > This behaviour doesn't happen when strategy changed to other than > 'linear'. > So why is Asterisk restart needed for this change? > Because the creation of the queue members container requires different properties for linear support. if (q->strategy == QUEUE_STRATEGY_LINEAR || q->strategy == QUEUE_STRATEGY_RRORDERED) /* linear strategy depends on order, so we have to place all members in a single bucket */ q->members = ao2_container_alloc(1, member_hash_fn, member_cmp_fn); else q->members = ao2_container_alloc(37, member_hash_fn, member_cmp_fn); The other strategies will work with the linear version of the container but are just not as efficient. Richard -- _ -- 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] Option 'd' of application Dial not working in 1.8.8-rc2
> > Hi, > > > > in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key > > while > > in ringing state puts the call to an one digit extension. > > > > In asterisk 1.8.8-rc2 this is not working anymore. After doing a > > diff > > on the code it seems to me, that in version 1.8.7 there is an > > autoanswer in application dial in case there is option 'd' present. > > Putting a "Answer" > > in the dialplan in front of the Dial-Statement solves the problem. > > > > Is this a bug or a feature? > > > > Thanks, > > > It was a necessary change. That automatic answer in the dial > application > broke DTMF attended transfer. See v1.8 SVN commit log -r336658. > > Will this affect 10.X or is it just a 1.8 path? Yes. It is in the 1.8, 10, and trunk branches. It was documented in the UPGRADE.txt file: * The Dial application d and H options do not automatically answer the call anymore. It broke DTMF attended transfers. Since many SIP and ISDN phones cannot send DTMF before a call is connected, you need to answer the call leg to those phones before using Dial with these options for them to have any effect before the dialed party answers. Richard -- _ -- 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] Option 'd' of application Dial not working in 1.8.8-rc2
> Hi, > > in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key while > in ringing state puts the call to an one digit extension. > > In asterisk 1.8.8-rc2 this is not working anymore. After doing a diff > on > the code it seems to me, that in version 1.8.7 there is an autoanswer > in > application dial in case there is option 'd' present. Putting a > "Answer" > in the dialplan in front of the Dial-Statement solves the problem. > > Is this a bug or a feature? > > Thanks, > It was a necessary change. That automatic answer in the dial application broke DTMF attended transfer. See v1.8 SVN commit log -r336658. Richard -- _ -- 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] Sangoma Card with 16E1 SS7 signaling
> > > Hi Team, > > > > > > i have been facing issues with sangoma card with 16 E1. > > > used LibSS7 > > > asterisk 1.6 > > > > > > with 8 E1 the links are stable , but moment i add another card of 8 > > > E1 > > > for to support 16 E1. link keeps fluctuating > > > > > > any idea why ? > > > > > Your 16th channel may be mismatched with the network. Timeslot 16 > > is usually used for signaling. channels => 1-15,17 > > The link is up on 16 th channel. My objective is to have 16 E1 to be > configure on single machine with two 8 port sangoma card. Which is > problem I am facing. Please let me know if you have any solution. > Sounds like it could be a clocking issue between the two cards then. Everything needs to use the same clock source. Richard -- _ -- 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] Sangoma Card with 16E1 SS7 signaling
> Hi Team, > > i have been facing issues with sangoma card with 16 E1. > used LibSS7 > asterisk 1.6 > > with 8 E1 the links are stable , but moment i add another card of 8 E1 > for to support 16 E1. link keeps fluctuating > > any idea why ? > Your 16th channel may be mismatched with the network. Timeslot 16 is usually used for signaling. channels => 1-15,17 Richard -- _ -- 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] Strange behavior over Zap chennels
Hi Doug, Thanks for the reply. Unfortunately I can't get my telco do do anything because I can't provide proof that is a problem with their lines. 2011/10/22 Doug Lytle > > Richard Reina wrote: > >> I have a server that is hooked to a channel bank (Adit 600). It has eight >> lines coming from a T1 through this adit's FXO card. >> > > I have a very similar setup, but my Adit 600 is used to provide FAX only. > Voice calls are PRI. And, my system hasn't been rebooted in over 2 years. > It's been a while since I've logged into my Adit, but I believe it has > logging. > > The usual suspects for analog lines are bad cable, water in the telco > equipment, Thunder storms. > > Doug > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > -- > __**__**_ > -- 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<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] Strange behavior over Zap chennels
I have a server that is hooked to a channel bank (Adit 600). It has eight lines coming from a T1 through this adit's FXO card. This particular * server is used primarily as a PBX and is not connected to the internet and has worked fine since 2006. However recently occasional calls are dropped with an Ahhn, Ahhn, Ahhn sound -- the kind of sound you hear if you leave the receiver of POTS phone off the look for too long. I have switched out the channel bank and it still happens. If I keep the CLI open I don't see any errors when this occurs -- however I can't be positive about this as it all happens very fast. I have also had my telco (XO Communications) out here several times. They say there is nothing wrong with the lines and that it has to be "my hardware" as the circuit is completing as we are able to talk briefly with the caller before the call is dropped. Nothing has been changed at all in the * servers configuration. The server has remained untouched and has not as much as been rebooted in a couple years. Has anyone ever heard of a similar problem and can anyone suggest how I might fix or diagnose it? Thank you very much for your time, Richard -- _ -- 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] Any help with these error messages???
> > [trunkgroups] > > > > [channels] > > > > [my-phones](!) > > usecallerid = yes > > hidecallerid = no > > callwaiting = yes > > usecallingpres = yes > > callwaitingcallerid = yes > > threewaycalling = yes > > transfer = yes > > canpark = yes > > cancallforward = yes > > callreturn = yes > > echocancel = yes > > echocancelwhenbridged = yes > > relaxdtmf = yes > > rxgain = 0.0 > > txgain = 0.0 > > group = 1 > > callgroup = 1 > > pickupgroup = 1 > > immediate = no > > > > context = my-phones > > signalling = fxo_ks > > > > [phone1](my-phones) > > signalling = fxs_ks > > callerid = "Andrew F Robinson" <(503)543-2338> > > dahdichan = 1 > > > > [phone2](my-phones) > > signalling = fxs_ks > > callerid = "Michael C Robinson" <(503)987-1322> > > dahdichan = 2 > > > > [phone3](my-phones) > > callerid = "2010" <2010> > > dahdichan = 3 > > > > [phone4](my-phones) > > callerid = "2011" <2011> > > dahdichan = 4 > > > > I don't see anywhere in the above file that I deal with pseudo. > > > > > It looks like you are attempting to manually configure the pseudo > > > channel > > > multiple times in chan_dahdi.conf. You do not need to explicitly > > > configure > > > the pseudo channel. The pseudo channel is always created and has > > > no > > > settable configuration parameters as far as I know. > > > > Please create an issue on the issue tracker: > https://issues.asterisk.org/jira > > With the above chan_dahdi.conf and indicate that it is generating > these > warnings when Asterisk loads: > [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Attempt to configure > channel -2 with signaling Unknown signalling -1 ignored because it is > already configured to be Pseudo. > Never mind. The chan_dahdi warnings about the Pseudo channel was already fixed in -r331955 of the v1.8 SVN branch and is in v1.8.7. Richard -- _ -- 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] G729 and Dahdi: Inbound forcing ulaw!
> Upgrade to 1.8.7.1 There was a bug fixed recently (I think in 1.8.6, > but might have been 1.8.7) which caused Asterisk to sometimes not > transcode when it should. A regression introduced in v1.8.7 broke the ability of the ./configure script to generate the HAVE_PRI_xxx defines for ISDN. Fix committed to v1.8 branch with -r339719. It is fixed in v1.8.8-rc2. You could simply use the ./configure script from v1.8.6. Richard > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JT > Sent: Wednesday, October 19, 2011 3:45 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] G729 and Dahdi: Inbound forcing ulaw! > > Happy day to ya list! > > I've recently deployed an update on our Asterisk server, taking it > from 1.2 up to 1.8.5 - going from zaptel to dahdi. Excitement levels > are high and performance so far is wonderful. > > The issue I have is I've purchased several G729 licenses, registered > them, installed the module in asterisk (per instructions) and for most > calls they work wonderfully. > > Inbound calls through the PRI however are forcing ulaw as the codec! I > noticed this as all of my phones are currently set to Disallow:All, > Allow:G729,gsm. (This was the same configuration with Asterisk > 1.2/Zaptel). When an inbound call arrives my logs fill up with: > chan_sip.c: Asked to transmit frame type ulaw, while native formats is > 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729) > > These did not appear with Asterisk 1.2/Zaptel. I have reviewed the > Dahdi settings and have found nothing relating to the codec being > passed or how to force it to use the codecs I've purchased. > > Any suggestions on where this might be set for inbound calls? > (Outbound calls, SIP to SIP and SIP to IAX2 to SIP are all using G729 > - this only relates to an inbound call from an external land line). -- _ -- 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] Any help with these error messages???
> [trunkgroups] > > [channels] > > [my-phones](!) > usecallerid = yes > hidecallerid = no > callwaiting = yes > usecallingpres = yes > callwaitingcallerid = yes > threewaycalling = yes > transfer = yes > canpark = yes > cancallforward = yes > callreturn = yes > echocancel = yes > echocancelwhenbridged = yes > relaxdtmf = yes > rxgain = 0.0 > txgain = 0.0 > group = 1 > callgroup = 1 > pickupgroup = 1 > immediate = no > > context = my-phones > signalling = fxo_ks > > [phone1](my-phones) > signalling = fxs_ks > callerid = "Andrew F Robinson" <(503)543-2338> > dahdichan = 1 > > [phone2](my-phones) > signalling = fxs_ks > callerid = "Michael C Robinson" <(503)987-1322> > dahdichan = 2 > > [phone3](my-phones) > callerid = "2010" <2010> > dahdichan = 3 > > [phone4](my-phones) > callerid = "2011" <2011> > dahdichan = 4 > > I don't see anywhere in the above file that I deal with pseudo. > > > It looks like you are attempting to manually configure the pseudo > > channel > > multiple times in chan_dahdi.conf. You do not need to explicitly > > configure > > the pseudo channel. The pseudo channel is always created and has no > > settable configuration parameters as far as I know. > > Please create an issue on the issue tracker: https://issues.asterisk.org/jira With the above chan_dahdi.conf and indicate that it is generating these warnings when Asterisk loads: [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Attempt to configure channel -2 with signaling Unknown signalling -1 ignored because it is already configured to be Pseudo. Richard -- _ -- 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] Any help with these error messages???
> On 11-10-16 01:51 AM, Michael C. Robinson wrote: > > [Oct 15 22:44:31] ERROR[29013] res_config_pgsql.c: PostgreSQL > > RealTime: > > Failed to connect database asterisk on 127.0.0.1: > > [Oct 15 22:44:31] WARNING[29013] res_config_pgsql.c: PostgreSQL > > RealTime: Couldn't establish connection. Check debug. > > [Oct 15 22:44:31] ERROR[29013] res_config_ldap.c: No directory URL > > or > > host found. > > [Oct 15 22:44:31] ERROR[29013] res_config_ldap.c: Cannot load LDAP > > RealTime driver. > > [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Attempt to configure > > channel -2 with signaling Unknown signalling -1 ignored because it > > is > > already configured to be Pseudo. > > [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Attempt to configure > > channel -2 with signaling Unknown signalling -1 ignored because it > > is > > already configured to be Pseudo. > > [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Attempt to configure > > channel -2 with signaling Unknown signalling -1 ignored because it > > is > > already configured to be Pseudo. > > [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Attempt to configure > > channel -2 with signaling Unknown signalling -1 ignored because it > > is > > already configured to be Pseudo. > > [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Ignoring any changes > > to > > 'userbase' (on reload) at line 23. > > [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Ignoring any changes > > to > > 'vmsecret' (on reload) at line 31. > > [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Ignoring any changes > > to > > 'hassip' (on reload) at line 35. > > [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Ignoring any changes > > to > > 'hasiax' (on reload) at line 39. > > [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Ignoring any changes > > to > > 'hasmanager' (on reload) at line 47. > > [Oct 15 22:44:32] WARNING[29013] cel_pgsql.c: CEL pgsql config file > > missing global section. > > [Oct 15 22:44:33] ERROR[29013] ais/clm.c: Could not initialize > > cluster > > membership service: Try Again > > > > I'm too much of a newbie to use a database to hold the configuration > > files and besides that I want to manage the files manually until I > > know what I'm doing in a gui/database environment. I don't want to > > set > > up LDAP either at this time, I'm not sure it would give me anything. > > I'm not sure what the last error pertains to, again I probably need > > to > > shut something off. I'm running Asterisk 1.8.3. I'm concerned about > > the signaling errors and not sure what is causing them as my > > chan_dahdi.conf appears to be correct. > > > WARNING != ERROR so you should be fine with chan_dahdi.so > > As for the other modules, if you are not using them add > > noload => res_config_pgsql.so > noload => res_config_ldap.so > > into your modules.conf > It looks like you are attempting to manually configure the pseudo channel multiple times in chan_dahdi.conf. You do not need to explicitly configure the pseudo channel. The pseudo channel is always created and has no settable configuration parameters as far as I know. Richard -- _ -- 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] failed to extend from 512 to 676
> I've got Asterisk 1.6.2.9 with RT SIP running, and for the most part, > it's working great. > > However, once in a while, I'll get some strange output from "sip show > peer .." > > For example: > === > *CLI> sip show peer 687F74D9848C-1 > * Name : 687F74D9848C-1 > Realtime peer: Yes, cached > Secret : > MD5Secret : > > Transfer mode: open > CallingPres : Presentation Allowed, Not Screened > Callgroup : > Pickupgroup : > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > === > > I'm not able to figure out what causes this and usually, I have to > prune the client, or reload sip to make it stop. > > Can someone tell me what causes it and how to fix it? > An Asterisk dynamic string (struct ast_str) needed more room for a string and failed to get it. There are many reasons why this might happen. I cannot say more without digging further into the code. Richard -- _ -- 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] Cisco AS5400XM
On Thu, Oct 6, 2011 at 11:25 AM, Kyle Sexton wrote: > I'm looking at the Cisco AS5400XM to convert some incoming T1s to SIP > signaling. Has anyone had any experience with these devices? The > feature cards that Cisco sells can be a little confusing. I'm > thinking something like below is what I need. > > (1) AS5400XM, AS5400XM Starter Kit (inc Chassis, MB, Def Mem) > (1) AS54-AC-RPS-PWR, AS5400 AC Redundant Power Supply > (1) AS54-DFC-8CT1, AS5400 OCTAL T1/PRI DFC Card > (2) AS54-DFC-108NP, AS5400 108 Voice/Universal Port Feature Card > > Any thoughts would be appreciated. Thanks. > We also looked it one time. One of the model is no longer supported by cisco. The replacement model with all DSP loaded was quite expensive, about $20k per box even on ebay. -- _ -- 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 PRI hangup
Ref: len= 1 (reference 93/0x5D) (Originator)
> < Message type: STATUS ENQUIRY (117)
> YYY Here we get reset YYY
> > Protocol Discriminator: Q.931 (8) len=7
> > Call Ref: len= 1 (reference 93/0x5D) (Terminator)
> > Message type: STATUS (125)
> > [14 01 00]
> > Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call
> state: Null (0)
> -- Got reject requesting packet 0... Retransmitting.
> < Protocol Discriminator: Q.931 (8) len=4
> < Call Ref: len= 1 (reference 93/0x5D) (Originator)
> < Message type: STATUS ENQUIRY (117)
> YYY Here we get reset YYY
> > Protocol Discriminator: Q.931 (8) len=7
> > Call Ref: len= 1 (reference 93/0x5D) (Terminator)
> > Message type: STATUS (125)
> > [14 01 00]
> > Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call
> state: Null (0)
> < Protocol Discriminator: Q.931 (8) len=8
> < Call Ref: len= 1 (reference 93/0x5D) (Originator)
> < Message type: DISCONNECT (69)
> < [08 02 82 a9]
> < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
> Location: Public network serving the local user (2)
> < Ext: 1 Cause: Temporary failure (41), class =
> Network Congestion (resource unavailable) (2) ]
> -- Processing IE 8 (cs0, Cause)
>
> At this point the call interrupts. And the log continues:
>
> q931.c:3826 q931_receive: call 93 on channel 1 enters state 12
> (Disconnect Indication)
> -- Channel 0/1, span 1 got hangup request, cause 41
> -- Executing [h@macro-dial:1] Macro("DAHDI/1-1", "hangupcall") in
> new stack
> -- Executing [s@macro-hangupcall:1] GotoIf("DAHDI/1-1", "1?skiprg")
> in new stack
> -- Goto (macro-hangupcall,s,4)
> -- Executing [s@macro-hangupcall:4] GotoIf("DAHDI/1-1",
> "0?skipblkvm") in new stack
> -- Executing [s@macro-hangupcall:5] NoOp("DAHDI/1-1", "Cleaning Up
> Block VM Flag: BLKVM/600/DAHDI/1-1") in new stack
> -- Executing [s@macro-hangupcall:6] NoOp("DAHDI/1-1", "Deleting:
> BLKVM/600/DAHDI/1-1 ") in new stack
> -- Executing [s@macro-hangupcall:7] GotoIf("DAHDI/1-1", "1?theend")
> in new stack
> -- Goto (macro-hangupcall,s,9)
> -- Executing [s@macro-hangupcall:9] Hangup("DAHDI/1-1", "") in new
> stack
> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
> 'DAHDI/1-1' in macro 'hangupcall'
> == Extension Changed 201[ext-local] new state Idle for Notify User 202
> == Extension Changed 201[ext-local] new state Idle for Notify User 215
> == Spawn extension (macro-dial, s, 7) exited non-zero on 'DAHDI/1-1'
> in macro 'dial'
> == Spawn extension (ext-group, 600, 17) exited non-zero on 'DAHDI/1-1'
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
> peerstate Disconnect Request
> q931.c:2967 q931_release: call 93 on channel 1 enters state 19
> (Release
> Request)
> > Protocol Discriminator: Q.931 (8) len=8
> > Call Ref: len= 1 (reference 93/0x5D) (Terminator)
> > Message type: RELEASE (77)
> > [08 02 81 a9]
> > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
> Location: Private network serving the local user (1)
> > Ext: 1 Cause: Temporary failure (41), class =
> Network Congestion (resource unavailable) (2) ]
> -- Hungup 'DAHDI/1-1'
> Timed out looking for release complete
> > Protocol Discriminator: Q.931 (8) len=8
> > Call Ref: len= 1 (reference 93/0x5D) (Terminator)
> > Message type: RELEASE (77)
> > [08 02 81 a9]
> > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
> Location: Private network serving the local user (1)
> > Ext: 1 Cause: Temporary failure (41), class =
> Network Congestion (resource unavailable) (2) ]
>
> What can be the cause of that problem?
>
> My config is: OpenSuSE with this packages:
>
> asterisk162-devel-1.6.2.17-83.1.i586
> asterisk162-dahdi-1.6.2.17-83.1.i586
> asterisk162-debuginfo-1.6.2.17-83.1.i586
> asterisk162-addons-1.6.2.3-38.2.i586
> asterisk162-1.6.2.17-83.1.i586
> dahdi-tools-devel-2.4.1-34.1.i586
> asterisk162-dahdi-1.6.2.17-83.1.i586
> dahdi-linux-kmp-default-2.4.1_2.6.31.14_0.6-14.1.i586
> dahdi-linux-2.4.1-14.1.i586
> dahdi-tools-2.4.1-34.1.i586
> free-pbx 2.7.0.10
> My dahdi-channels.conf is the following:
>
> group=0,11
> context=from-pstn
> switchtype = euroisdn
> signalling = bri_cpe_ptmp
> channel => 1-2
> context = default
> group = 63
>
> my chan_dahdi.conf is the following:
>
> [trunkgroups]
>
> [channels]
> context=incoming
> internationalprefix = 00
&
Re: [asterisk-users] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server
> Actually it doesn't say "AGI(async:script)" it says "AGI(async:agi)" > and than continues further to setting up an AMI user so the script is > executed through the manager interface?? Than it says > "AGI(agi:async)".?? Well most importantly it says "Cons of async AGI: > It is the most complex method of using AGI to implement." ..:) I have > been interested in Async AGI as well and after reading your post > looked into the link you provided, seems different than what we > immediately think, a background process. AGI(agi:async) is the correct invocation. AsyncAGI is still an AGI and behaves as such. The primary difference between the flavors of AGI is where the commands AGI executes come from. For AsyncAGI, the commands are passed to Asterisk with AMI actions. Command responses are passed back the same way. One thing to note is that the AGI AMI actions use "CommandID" instead of the normal "ActionID". > > Is there anyway for me to make asynchronous AGIs work? I've tried > > searching online to no avail. The most definitive documentation is always the source code. :) For AGI this is in the res/res_agi.c file. Richard -- _ -- 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] Native bridging to SIP endpoints on the same NAT'd network
Hi, I have the following setup: Asterisk <-> Nat <-> Internet <-> Nat <-> 2 x SIP endpoints With directmedia=no I can make a call between the two SIP endpoints; the RTP stream being passed through the Asterisk box. Obviously, this is sub-optimal. I attempted to enable bridging of the call between the 2 endpoints directly, given that they are on the same non-routeable private net. With directmedia=nonat, I see Asterisk report the bridging of the calls but both sides of the call are routed to the originating endpoint so effectively, the call becomes an echo-loop. There is no audio on the second end-point although the call remains up. I assume this is some sort of firewall/nat/routing issue. Could someone explain what is possibly going on and perhaps offer a solution? Cheers, Richard. -- _ -- 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] RTP stream when * and Xlite are both behind Nat devices.
Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp stream to be sent to the public facing address of the softphone? Cheers, Richard. -- _ -- 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] Unexpected behavior change from Asterisk 1.6.2.14 to Asterisk 1.8.5.0
s returing BUSY instead of > CHANUNAVAIL when the particular channel that was tried is in use by a > different call. We worked around the issue by applying the > recommendation suggested in the ticket (create DAHDI groups in > chan_dahdi.conf and use these as trunks). However, I believe the > previous behavior was correct and the new behavior to be in error. The > workaround suggested by the ticket will not work in a scenario where a > DAHDI group has all of its channels busy with calls, and the > administrator intends additional calls to be routed > through non-DAHDI trunks (such as SIP/IAX trunks or custom trunks). > > My questions: > > Is the new behavior the intended one? > If the new behavior is intentional, then how should I set up an > scenario in which calls will be routed through SIP when all DAHDI > channels are in use, yet will not try to route calls through SIP when > the destination is truly busy? > If the new behavior is a bug and not intentional, at what level should > I look for the problem? At Asterisk, or at the driver level? The card > is an OpenVox card (opvxa1200) for which source code is available. > I think the new behavior is a bug. It is most likely in chan_dahdi.c:dahdi_request() when it finds that the requested channel or no channels in the group are available. Richard -- _ -- 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] Help with pri call giving error.
- Original Message - > I am not getting calls going out my PRI. > I am getting an error condition. > > There are no errors in /var/log/asterisk/messages. > > more /etc/dahdi/system.conf > loadzone=us > defaultzone=us > span=1,1,0,esf,b8zs > bchan=1-2 > dchan=24 > echocanceller=mg2,1-2 > > > more /etc/asterisk/chan_dahdi.conf > [channels] > pridialplan=unknown > prilocaldialplan=unknown > switchtype=national > signalling=pri_net > relaxdtmf=yes > echocancel=yes > echocancelwhenbridged=yes > echotraining=400 > callerid=asreceived > context=smvoice-incoming > group=1 > channel => 1-2 > > > and pri logging shows: > > > DL-DATA request > > Protocol Discriminator: Q.931 (8) len=64 > > TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent from originator) > > Message Type: SETUP (5) > TEI=0 Transmitting N(S)=4, window is open V(A)=4 K=7 > > > Protocol Discriminator: Q.931 (8) len=64 > > TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent from originator) > > Message Type: SETUP (5) > > [04 03 80 90 a2] > > Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer > > capability: Speech (0) > > Ext: 1 Trans mode/rate: 64kbps, > > circuit-mode (16) > > User information layer 1: u-Law (34) > > [18 03 a9 83 81] > > Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 > > Exclusive Dchan: 0 > >ChanSel: As indicated in following octets > >Ext: 1 Coding: 0 Number Specified Channel > >Type: 3 > >Ext: 1 Channel: 1 Type: NET] > > [1e 02 80 83] > > [Kcsdbsigns*CLI> > [0K> Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard > (0) 0: 0 Location: User (0) > >Ext: 1 Progress Description: Calling > >equipment is non-ISDN. (3) ] > > [28 17 b1 41 64 6d 69 6e 20 53 79 73 74 65 6d 20 53 59 53 20 41 44 > > 4d 49 4e] > > Display (len=23) Charset: 31 [ Admin System SYS ADMIN ] > > [6c 03 00 81 30] > > Calling Number (len= 5) [ Ext: 0 TON: Unknown Number Type (0) NPI: > > Unknown Number Plan (0) > >Presentation: Presentation permitted, > >user number passed network screening (1) > >'0' ] > > [70 0d 80 39 31 33 31 37 35 30 36 38 30 31 32] > > Called Number (len=15) [ Ext: 1 TON: Unknown Number Type (0) NPI: > > Unknown Number Plan (0) '913175551212' ] > > [Kcsdbsigns*CLI> > [0Kq931.c:6036 q931_setup: Call 32771 enters state 1 (Call Initiated). > Hold state: Idle > > [Kcsdbsigns*CLI> > [0K == Manager 'MessageNet' logged off from 127.0.0.1 > > [Kcsdbsigns*CLI> > [0K > < Protocol Discriminator: Q.931 (8) len=9 > < TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent to originator) > < Message Type: RELEASE COMPLETE (90) > < [08 02 81 81] > < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 > Location: Private network serving the local user (1) > < Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal > Event (0) ] > Received message for call 0x1b0f7f40 on link 0x1b0f08b0 TEI/SAPI 0/0 > -- Processing IE 8 (cs0, Cause) > q931.c:8567 post_handle_q931_message: Call 32771 enters state 0 > (Null). Hold state: Idle > -- Channel 0/1, span 1 got hangup, cause 1 > > > > > Why is the call failing? > The cause code says Unallocated (unassigned) number. You are dialing an invalid number. Is the 9 supposed to be in your called number? Richard -- _ -- 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] Custom Dialplan
Hallo Barry, extensions_additional.conf is supposed to be edited by FreePBX. Gopal, on using extensions_custom, the SIP phones work however the details are not captured in the reporting mechanism of FreePBX, which is what I need most. Richard Zulu Twitter www.twitter.com/richardzulu http://www.linkedin.com/in/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* On Mon, Aug 8, 2011 at 2:03 PM, Gopal krishnan wrote: > Use extensions_custom.conf file to update your custom configurations. > > > On Sun, Aug 7, 2011 at 3:59 AM, Barry L. Kline wrote: > >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> On 08/05/2011 04:32 AM, Richard Zulu wrote: >> >> > I would like to import my dialplan into freepbx+asterisk since I am >> > switching to that...how can I create my own custom dialplan in >> > freepbx? >> >> I'm not sure why you'd want to... freepbx is anathema to custom >> dialplans. That said, I believe you end up naming your >> "extensions.conf" file to "extensions_additional.conf" and freepbx will >> pick it up when it starts. >> >> It's been a long, long time since I've dealt with freepbx -- in fact I >> went the other way: from freepbx+asterisk to pure asterisk. When I was >> using freepbx that was the solution you seek. >> >> Barry >> >> -BEGIN PGP SIGNATURE- >> Version: GnuPG v1.4.5 (GNU/Linux) >> >> iD8DBQFOPcAxCFu3bIiwtTARAkjKAKCPCgcoaRyPNs7BXhge7xxcy7C2qQCdF6hx >> 2Bwz/YEUSbKFsfzD9V0xX6Q= >> =W2Dn >> -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 >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk reporting
Hallo, I have a production asterisk server running on Ubuntu however all my configs where done using the CLI. I would like to implement a reporting element into the server so I can know the number of calls made, for what duration, on what dates. What tool can I use that can fit within any already laid out dialplan? Thanks Richard Zulu Twitter www.twitter.com/richardzulu http://www.linkedin.com/in/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* On Thu, Aug 11, 2011 at 4:12 AM, neo haux wrote: > Hi > > I want to change my old answering phone machine and two wireless phones > with asterisk box + degium TDM400p (3 fxs+1FXO)+ one LAN phone(Aastra Nortel > 9133i) + Wifi/SIP phone > > I am wondering if I´ll lost actual functionalities that are present in my > old answering machine: > 1) is it possible to show the caller number (coming from PSTN/FXO) in both > SIP phones (wifi/SIP and LAN phone) ? Does SIP protocol take in charge this > functionality > > 2) Most important question is : can I see on those internal phones > (Wifi/SIP phone and LAN phone) that I´ve some recoded messages on asterisk. > Indeed, I have this fucntionality with my old answering machine where I can > see the number of new messages recorded in a big LCD screen. > > > Thx > > > -- > _ > -- 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] ASterisk is Going stop whenever restart the server
> I am using goautodial, I am using 20channels telcom PRI line and in my > server DIgium TE120 PRI card which is for 31 channel. with this > configuration > I am able to call from server . but problem whenever i restarted the > server that time is Asterisk is stop then I am not able to call > outside. > how to resolve this issue. every time whenever restart the server this > asterisk it going stop. I don't know what you mean by the stop references. If you mean the call fails, then that could be because the call attempted to use a channel not provisioned for your PRI line. Your configuration provisions all 30 channels for an E1 line when you said you have 20 channels. > > inthis server my hardware configuration is > /etc/asterisk/dahdi-channels.conf > group=0,11 > context=default > switchtype = euroisdn > signalling = pri_cpe > channel => 1-15,17-31 ^^^ This line is creating all of the channels for an E1 so chan_dahdi thinks it has all 30 channels available. You said you only have 20 channels available so the channels line is wrong here. It should be something like channel => 1-15,17-21 Also anything after a channel line does not apply to the channels created by a channel line. > context = default > group = 63 > > /etc/asterisk/chan_dahdi.conf > [trunkgroups] > > [channels] > #include dahdi-channels.conf > language=en > context=default > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > canpark=yes > restictcid=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=yes > echotraining=800 > relaxdtmf=yes > rxgain=0.0 > txgain=0.0 > ;group=1 > ;callgroup=1 > ;pickupgroup=1 > busydetect=yes > busycount=6 > immediate=no > resetinterval=never > switchtype=euroisdn > signalling=pri_cpe > pridialplan=unknown > prilocaldialplan=unknown > group=0 > channel => 1-20 ^^^ This channel line is wrong. It attempts to redefine channels you have already defined earlier by the dahdi-channels.conf include. You should have gotten warning messages when Asterisk loaded about channels already defined. > > > /etc/dahdi/system.conf > span=1,1,0,ccs,hdb3,crc4 > # termtype: te > bchan=1-15,17-31 > dchan=16 > echocanceller=mg2,1-15,17-31 Richard -- _ -- 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] Custom Dialplan
Hey, I have been using asterisk on slackware and had thus come up with my own dialplan. I would like to import my dialplan into freepbx+asterisk since I am switching to that...how can I create my own custom dialplan in freepbx? Thanks Richard Zulu Twitter www.twitter.com/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* -- _ -- 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] Segmentation Fault
Hey, I have installed Asterisk 1.8 on slackware 13.1, php, mysql and apache. I am later to install freepbx to help with reporting on VOIP activity. However, after installing asterisk, I am getting a segment fault. My log file shows this: darkstar kernel: asterisk[2660]: segfault at 81c4f ip 77514810 sp 7fffcd48 error 4 in libc-2.11.1.so[77492000+16b000] I have used gdb so that I can perform a backtrace however the program executes and exits without a stack thus not helpful. Any help is appreciated! Richard Zulu Twitter www.twitter.com/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* -- _ -- 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] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
> Can you please point me to the patch that you just made? > The patch is committed to v1.6.2 SVN branch. Patch for v1.6.2 only. r330490 | jrose | 2011-08-01 16:08:10 -0500 (Mon, 01 Aug 2011) | 12 lines Asterisk 18103 - Fix reload crash caused by destroying default parking lot Default parking lot was being destroyed in reload and was not being rebuilt properly. This patch keeps features.c reload from destroying the default parking lot in 1.6.2. Bug was caused by a hasty backport which didn't test reload enough times to catch the problem. (closes issue ASTERISK-18103) Reported by: 808blogger Review: https://reviewboard.asterisk.org/r/1337/ Also -r330505 to fix a ref leak with the above patch. Richard -- _ -- 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] Capturing call Reject/Decline events on a PRI line
> There is no event for Asterisk to recognize. The PROGRESS message just > says that there is an audio message available for the caller to listen > to. Asterisk just passes the indication to the peer channel and opens > the audio path. It is the caller who must recognize any audio message > that their call has been dropped. > > Thanks for the explanation. Any suggestion on how to recognise that > the call has been dropped? > > > As far as ISDN is concerned, the > call has not been answered yet so Asterisk must keep waiting. > As far as the ISDN signaling is concerned, the call is still going. There is no signaling to indicate the call is not going to proceed any further. Richard -- _ -- 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] Capturing call Reject/Decline events on a PRI line
> We enable pri intense debug with the standard asterisk PRI dialplan, > collected the logs and you can find the logs attached to the mail. > > After the call was made, the called party cut the call, and asterisk > doesn't seem to recognise the event. > > I can't make much sense of the logs given my non-existent background > in telephony. Would somebody here help me figure why the event wasn't > captured? > There is no event for Asterisk to recognize. The PROGRESS message just says that there is an audio message available for the caller to listen to. Asterisk just passes the indication to the peer channel and opens the audio path. It is the caller who must recognize any audio message that their call has been dropped. As far as ISDN is concerned, the call has not been answered yet so Asterisk must keep waiting. Richard -- _ -- 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] Securing Asterisk
> Can please the Powers that Be reconsider and add this option to sip.conf? What "Powers that Be"? This is open-source software! If you need an option in sip.conf, just add it! -- _ -- 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] callgroup and pickupgroup (Carlos Chavez)
> What I am looking for is: > > If my IP Phone is related to a pickup group #1 and a call is ringing > at pickup group #2, so I can pickup the call that is ringing at group > #2 and I do not know its extensions? > > In other words, how can I pickup any phone that is ringing without > knowing its extension, but this Phone who is ringing, it is belongs to > ther pickup group that my phone is belongs to. > > This feature is required when we have two floors, and we divided first > floor to be in a pickup group and second floor to be in a second > pickup group, so if a call is ringing in floor 2 and I am in floor 1 > (and I hear the ring), so how I will pickup the call that is in floor > 2 and I do not know at which extension is ringing. > > One more thing: what is the use for the callgroup? Does it has any use > related to pickup the call? pickupgroup uses callgroup. callgroup is the extension groups that an extension belongs to. pickupgroup is the call groups that an extension can pickup. Both callgroup and pickupgroup are internally mapped to a 64 bit integer. This is why you can only have groups 0 to 63. Look at the various sample conf files. Configuration options are generally documented there. Richard -- _ -- 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] Redirecting call from one E1 to another?
> In article <296076780.5348.1310743930593.JavaMail.root@zimbra>, > Richard Mudgett wrote: > > > I would suggest Two B-Channel Transfer (TBCT), transferring to a > > > unique > > > number (received as DNIS by the other server) that would identify > > > the > > > call > > > as transferred from the first server and, perhaps, the reason for > > > the > > > transfer. > > > > > > It looks like TBCT may not have been implemented in Asterisk for > > > EuroISDN. > > > > > It is implemented in EuroISDN(ETSI). > > You need Asterisk 1.8 and libpri 1.4.12 to take advantage of the > > feature. > > Thanks. I had just a moment ago found the ETSI spec and then the > libpri > file rose_etsi_ect.c, so I was almost there! > > Does Asterisk implement actioning an ECT from the remote end (so as to > test with back-to-back Asterisks), or just support generating an ECT? > Yes, it does both ends. ETSI is the only switchtype that libpri implements initiating ECT and accepting ECT. Asterisk can initiate ECT to eliminate a tromboned call if the call is natively bridged and chan_dahdi.conf transfer=yes is configured. Richard -- _ -- 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] Redirecting call from one E1 to another?
> I would suggest Two B-Channel Transfer (TBCT), transferring to a > unique > number (received as DNIS by the other server) that would identify the > call > as transferred from the first server and, perhaps, the reason for the > transfer. > > It looks like TBCT may not have been implemented in Asterisk for > EuroISDN. > It is implemented in EuroISDN(ETSI). You need Asterisk 1.8 and libpri 1.4.12 to take advantage of the feature. Richard -- _ -- 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] Mysterious dropped calls
> So I'm now using asterisk 1.8.5rc1 for Asterisk. I'm still getting > mysterious dropped calls. This only happens on calls that are outbound > on Dahdi and mostly happens in conference calls particularly > 8xx-xxx- > > This is the output of the hangup. > > [Ksebpbx1*CLI> > [0KPRI Span: 1 q931_hangup: other hangup > PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, > peerstate Connect Request, hold-state Idle > PRI Span: 1 q931.c:4845 q931_disconnect: Call 32985 enters state 11 > (Disconnect Request). Hold state: Idle > PRI Span: 1 > PRI Span: 1 > DL-DATA request > PRI Span: 1 > Protocol Discriminator: Q.931 (8) len=9 > PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from > originator) > PRI Span: 1 > Message Type: DISCONNECT (69) > PRI Span: 1 TEI=0 Transmitting N(S)=51, window is open V(A)=51 K=7 > PRI Span: 1 > PRI Span: 1 > Protocol Discriminator: Q.931 (8) len=9 > PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from > originator) > PRI Span: 1 > Message Type: DISCONNECT (69) > PRI Span: 1 > [08 02 81 90] > PRI Span: 1 > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) > Spare: 0 Location: Private network serving the local user (1) > PRI Span: 1 > Ext: 1 Cause: Normal Clearing (16), class = Normal Event > (1) ] > -- Hungup 'DAHDI/i1/18662006965-1be' > > [Ksebpbx1*CLI> > [0K == Spawn extension (from-sip, 18662006965, 1) exited non-zero on > 'SIP/7027-0520' > > [Ksebpbx1*CLI> > [0KPRI Span: 1 > PRI Span: 1 < Protocol Discriminator: Q.931 (8) len=5 > PRI Span: 1 < TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent to > originator) > > [Ksebpbx1*CLI> > [0KPRI Span: 1 < Message Type: RELEASE (77) > PRI Span: 1 Received message for call 0x20984f0 on 0x7f7f804f24d0 > TEI/SAPI 0/0, call->pri is 0x7f7f804f24d0 TEI/SAPI 0/0 > PRI Span: 1 q931.c:7237 post_handle_q931_message: Call 32985 enters > state 0 (Null). Hold state: Idle > Span: 1 Processing event: PRI_EVENT_HANGUP > PRI Span: 1 q931_hangup: other hangup > PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, > peerstate Release Request, hold-state Idle > PRI Span: 1 > PRI Span: 1 > DL-DATA request > PRI Span: 1 > Protocol Discriminator: Q.931 (8) len=9 > PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from > originator) > PRI Span: 1 > Message Type: RELEASE COMPLETE (90) > PRI Span: 1 TEI=0 Transmitting N(S)=52, window is open V(A)=52 K=7 > PRI Span: 1 > PRI Span: 1 > Protocol Discriminator: Q.931 (8) len=9 > PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from > originator) > PRI Span: 1 > Message Type: RELEASE COMPLETE (90) > PRI Span: 1 > [08 02 81 90] > PRI Span: 1 > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) > Spare: 0 Location: Private network serving the local user (1) > PRI Span: 1 > Ext: 1 Cause: Normal Clearing (16), class = Normal Event > (1) ] > PRI Span: 1 q931_hangup: other hangup > PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, > peerstate Null, hold-state Idle > PRI Span: 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, > peerstate Null, hold-state Idle > > Any ideas? The decision to drop the call within Asterisk has already been made and is not shown in the trace. This trace is just showing the clearing of the call being initiated by the Asterisk side with a cause of normal clearing. Nothing is unexpected here. You need to capture debug output of earlier events to figure this out. Richard -- _ -- 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] calleridname presentation Asterisk => Siemens
> Hi
>
>
> I change for first way in Asterisk 1.8:
>
>
>
> [teste]
> include=>rota00
> exten=>1504,1,Set(CALLERID(name-charset)=unknown)
> exten=>1504,2,Dial(DAHDI/g1/${EXTEN},60,tTwW)
> exten=>1504,3,Hangup()
>
>
> But, in debug of the span show the simple form:
>
>
>
> 1 namePresentationAllowedSimple Context Specific [0 0x00] =
> 1 <52 61 66 61 65 6C> - "Rafael"
>
Looks like the simple way is not going to work. Chan_dahdi/sig_pri
is still using the old API call that does not pass the character set
value to libpri.
You'll need to go with method two then and use the patch I put in
the previous response.
Richard
>
> 2011/7/1 Richard Mudgett < rmudg...@digium.com >
>
>
> > I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i
> > can't show the callerid name in the way Asterisk ==> Siemens. I
> > realized that Asterisk send calleridname in format
> > "namePresentationAllowedSimple" to Siemens e Siemens send
> > calleridname
> > in format "namePresentationAllowedExtended". I need change the
> > format
> > of the calleridname in asterisk.
> >
> >
> > How to change?
>
> There are two ways:
>
> 1) With Asterisk v1.8 change the character set of the name to
> CALLERID(name-charset)=unknown.
> The default character set of iso8859-1 uses the simple form since that
> is the default character set of the extended form.
>
> 2) Change libpri as follows in the function rose_enc_qsig_Name() to
> always send the extended form:
> --- rose_qsig_name.c (revision 2267)
> +++ rose_qsig_name.c (working copy)
> @@ -94,22 +94,12 @@
> /* Do not encode anything */
> break;
> case 1: /* presentation_allowed */
> - if (name->char_set == 1) {
> - ASN1_CALL(pos, asn1_enc_string_bin(pos, end,
> ASN1_CLASS_CONTEXT_SPECIFIC | 0,
> - name->data, name->length));
> - } else {
> - ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos, end,
> - ASN1_CLASS_CONTEXT_SPECIFIC | 1, name));
> - }
> + ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos, end,
> + ASN1_CLASS_CONTEXT_SPECIFIC | 1, name));
> break;
> case 2: /* presentation_restricted */
> - if (name->char_set == 1) {
> - ASN1_CALL(pos, asn1_enc_string_bin(pos, end,
> ASN1_CLASS_CONTEXT_SPECIFIC | 2,
> - name->data, name->length));
> - } else {
> - ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos, end,
> - ASN1_CLASS_CONTEXT_SPECIFIC | 3, name));
> - }
> + ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos, end,
> + ASN1_CLASS_CONTEXT_SPECIFIC | 3, name));
> break;
> case 3: /* presentation_restricted_null */
> ASN1_CALL(pos, asn1_enc_null(pos, end, ASN1_CLASS_CONTEXT_SPECIFIC |
> 7));
--
_
-- 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] calleridname presentation Asterisk => Siemens
> I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i
> can't show the callerid name in the way Asterisk ==> Siemens. I
> realized that Asterisk send calleridname in format
> "namePresentationAllowedSimple" to Siemens e Siemens send calleridname
> in format "namePresentationAllowedExtended". I need change the format
> of the calleridname in asterisk.
>
>
> How to change?
There are two ways:
1) With Asterisk v1.8 change the character set of the name to
CALLERID(name-charset)=unknown.
The default character set of iso8859-1 uses the simple form since that is the
default character set of the extended form.
2) Change libpri as follows in the function rose_enc_qsig_Name() to always send
the extended form:
--- rose_qsig_name.c(revision 2267)
+++ rose_qsig_name.c(working copy)
@@ -94,22 +94,12 @@
/* Do not encode anything */
break;
case 1: /* presentation_allowed */
- if (name->char_set == 1) {
- ASN1_CALL(pos, asn1_enc_string_bin(pos, end,
ASN1_CLASS_CONTEXT_SPECIFIC | 0,
- name->data, name->length));
- } else {
- ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos, end,
- ASN1_CLASS_CONTEXT_SPECIFIC | 1, name));
- }
+ ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos, end,
+ ASN1_CLASS_CONTEXT_SPECIFIC | 1, name));
break;
case 2: /* presentation_restricted */
- if (name->char_set == 1) {
- ASN1_CALL(pos, asn1_enc_string_bin(pos, end,
ASN1_CLASS_CONTEXT_SPECIFIC | 2,
- name->data, name->length));
- } else {
- ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos, end,
- ASN1_CLASS_CONTEXT_SPECIFIC | 3, name));
- }
+ ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos, end,
+ ASN1_CLASS_CONTEXT_SPECIFIC | 3, name));
break;
case 3: /* presentation_restricted_null */
ASN1_CALL(pos, asn1_enc_null(pos, end,
ASN1_CLASS_CONTEXT_SPECIFIC | 7));
Richard
--
_
-- 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 1.6 Dahdi on Centos 5.2
> Hello, I have Asterisk 1.6 running on Centos, Also I have one analog > telephone line coming on > Wildcard TDM400P REV E/F Board 5 > > I can't get asterisk to dectect call coming from analog line. > Here is my /etc/dahdi/system.conf > fxsks=1 > > # global data > loadzone = us > defaultzone = us > > > /etc/asterisk/chan_dahdi.conf > [channels] > language=en > context=my-phones > switchtype=national > signalling=fxs_ks > channel => 1 > > > /etc/asterisk/extensions.conf > [globals] > CONSOLE=DAHDI/1 > TRUNK=DAHDI/4 > TRUNKMSD=1 > > [my-phone] > exten => 2000,1,Dial(DAHDI/1/116) > exten => 2000,2,cONGESTION > > exten => 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline > exten => 2001,2,HangUp() > > exten => 1001,1,Dial(DAHDI/1/7608514114) > exten => 1001,2,HangUp() > > exten => ,1,Dial(DAHDI/1/7608514114) > exten => l111,2,HangUp() The context in chan_dahdi.conf is my-phones which differs from the my-phone context in extensions.conf. Richard -- _ -- 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] Monitor Asterisk and Ast-gui
Hey, I have installed asterisk 1.8 on Slackware 13.1 from source and it is working well. I have 300 ip phones in a natted environment and my asterisk server has a public IP I would love to monitor my SIP activity on my VOIP Server, statistics like amount of sip traffic, who made what call and to whom, how many calls were made in a month, how many ip phones are up and running, which sip phone has made most calls among others. How best can I do that? On the other hand, I have also tried installing ast-gui onto asterisk 1.8, it has installed well but it however keeps looping whenever i try to login in, it says checking permissions on gui folder and loops. Haven't found much help on other mailing lists, any direction given in welcome. Thanks Richard Zulu Twitter www.twitter.com/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* -- _ -- 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] Problem with detecting fax on PRI/DAHDI channels
> [trunkgroups]
>
> [channels]
> switchtype=qsig
> context = from-pstn
> group = 0
> signalling = pri_cpe
> channel = 1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124
Everything after the channel line above will have no effect on the
channels created by the above line. Thus the faxdetect=both below
will not have any effect.
>
> usecallingpres=yes
> sendcalleridafter = 2
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain = 0.0
> txgain = 0.0
> group=1
> callgroup=1
> pickupgroup=1
> faxdetect=both
>
> #include /etc/asterisk/dahdi-channels.conf
Please note that the include is part of the chan_dahdi.conf
config file.
>
> Sawan
>
> On Fri, Jun 24, 2011 at 12:15 AM, Eric Wieling
> wrote:
> >
> >> -Original Message-
> >> From: asterisk-users-boun...@lists.digium.com
> >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> >> Sawan Vithlani
> >> Sent: Thursday, June 23, 2011 10:12 AM
> >> To: asterisk-users@lists.digium.com
> >> Subject: [asterisk-users] Problem with detecting fax on
> >> PRI/DAHDI channels
> >>
> >> Hello all,
> >>
> >> I am using call files to dial out to a set of PSTN numbers.
> >> The calls are going out fine and being handled correctly by
> >> the dial plan.
> >>
> >> The problem occurs when I accidentally call a fax machine. I
> >> would expect the dial plan to pick this up and jump to the
> >> "fax" extension in the current context. This is not happening.
> >>
> >> In the call file I send the call out via
> >> Dial(DAHDI/g0/) and then it goes to my dial plan context:
> >>
> >> [SendCall]
> >> exten => start,1,Answer()
> >> Â Â same => n, Wait(5)
> >> Â Â same => n, . ;; do some more stuff
> >>
> >> exten => fax,1,Verbose(got a Fax on ${EXTEN}) same =>n, Hangup()
> >>
> >>
> >>
> >> Â I do get a message on the CLI telling me "Channel 63
> >> detected a CED tone from the network" but still no jumping to
> >> the fax extension.
> >> This is while dialplan is in the Wait(5).
> >>
> >> CLI> core show version
> >> Asterisk 1.8.4.1-1digium1~natty built by pbuilder @ nighthawk on a
> >> x86_64 running Linux on 2011-05-23 22:05:17 UTC
> >>
> >> ISDN lines connected via Digium TE412P card.
> >>
> >> I have "faxdetect = both" in chan_dahdi.conf in the general
> >> section as well as specifically for the configured spans.
> >
> > Show us your chan_dahdi.conf
Richard
--
_
-- 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] How to set BRI-to-BRI trunk using 2 HA8+B400M cards
> The link between the Asterisk boxes must not be terminated on both > ends. Termination resistors on both ends is practically guaranteed > to cause link issues. > > > Very interesting but I'm afraid I still don't get it. > > For various reasons, my goal is build an asterisk + patton solution > that will be later plugged into 8 BRIs like this (hoping my ASCII art > will work) : > > Patton x4 BRI PSTN > | | > | | > Asterisk1 x4 BRI Connections like this? Patton -- x4 BRI --- PSTN Asterisk1 --- x4 BRI --- PSTN Asterisk1 --- BRI?? Patton (Cross link between Asterisk1 and Patton (Beware of potential clock loop)) > > The box Asterisk2 ast I referred to it previously, is here to behave > as the PSTN. > As a consequence, I tought that : > 1. I should set each Asterisk2 port as NT/PTP. The NT/PTP port should not have termination resistors enabled. Richard -- _ -- 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] How to set BRI-to-BRI trunk using 2 HA8+B400M cards
> My setup is: > Asterisk1 w/ HA8+B400M --- Asterisk2 w/ HA8+B400M --- Patton SN4638 > > Asterisk1 is in TE/PtP (with termination) > Asterisk2 is in NT/PtP( with termination) > Patton is in TE/PtP > The cable between Asterisk boxes is an RJ11M-RJ11M (custom made with > pinouts 1 to 1, 2 to 2, ...). > The cable between Asterisk2 and Patton boxes is an RJ11M-RJ45M from > Digium package > > > > At the moment, I can't get a stable trunk (as described above) between > 2 Asterisk boxes but I can get a stable one betwen Asterisk2 and > Patton boxes. > > Once I could once pass a phone between both boxes but most of the > time, the trunk is down : > asterisk -rx "pri show spans" > PRI span 1/0: Provisioned, In Alarm, Down, Active > ... > or > ... > asterisk -rx "pri show spans" > PRI span 1/0: Provisioned, Up, Active > PRI span 2/0: Provisioned, In Alarm, Down, Active > > > > In Asterisk1 (the TE/PtP box), config is : > /etc/dahdi/system.conf > span=1,1,0,ccs,ami,term,te > bchan=1-2 > hardhdlc=3 > echocanceller=oslec,1-2 > > /etc/asterisk/dahdi-channels.conf > ; Span 1: WCBRI/0/0 "HA8- Board 1" (MASTER) AMI/CCS RED > group=1,11 > context=remote > switchtype = euroisdn > signalling = bri_cpe > channel => 1-2 > context = default > group = 63 > > > > In Asterisk2 (the NT/PtP box), config is : > /etc/dahdi/system.conf > span=2,0,0,ccs,ami,term,nt > # termtype: nt > bchan=4-5 > hardhdlc=6 > echocanceller=oslec,4-5 > > > /etc/asterisk/dahdi-channels.conf > ; Span 2: WCBRI/0/1 "HA8- Board 1" > group=1,12 > context=remote > switchtype = euroisdn > signalling = bri_net > channel => 4-5 > context = default > group = 63 > > > Please, take note that : > A. port 1 in Asterisk1 is connected to port2 in Asterisk2. > B. port 1 in Asterisk2 is connected to a Patton 4638 port. > C. ports 1 and 2 in Asterisk2 belongs to the same group (using 2 > different groups doesn't change anything, it seems but I didn't digg > much). > D. I'm using asterisk 1.6.1.18, libpri 1.4.11.5 and dahdi 2.4.1(.2) > > > How can I make this work ? > Suggestions The link between the Asterisk boxes must not be terminated on both ends. Termination resistors on both ends is practically guaranteed to cause link issues. Also make sure that clocking is only supplied by one side of the link. (The Asterisk1 clock should be slaved to Asterisk2 which should be in slaved to the Patton link.) Richard -- _ -- 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] how to get on hold events with AMI
> I am getting events using asterisk 1.4.41. However, when I place a > call > on hold I do not get that event. > > Some of the events I am getting I show below. I wish to monitor when > channels are placed on hold > and taken off hold. Set callevents=yes in sip.conf to enable Hold/Unhold AMI call events for SIP channels. Richard -- _ -- 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] "pickupsound = beep" kills call pickup in Asterisk 1.8.4.2
> I have discovered that if I enable pickupsound = beep in > features.conf, > if I try to do a pickup with *8, the calling channel keeps on ringing, > while the phone where I pick-up from shows that the call has been > answered (I don't know where though). Also, it seems to completely > bugger up my outgoing IAX trunk (I really can't see the connection, as > I'm doing pick-up for a SIP channel). I can only shut Asterisk down > with > killall asterisk -s9 - nothing else works. > > I've tried starting the console with asterisk -rvv - but there is > nothing unusual there. > > Could someone please confirm this behaviour on their box, before I go > and submit a bug - in case I am doing something wrong? > > As soon as I comment out "pickupsound = beep" - everything works just > fine and I can do call pickup with *8. > > Sebastian I think this problem is already fixed in the SVN 1.8 branch. See https://issues.asterisk.org/jira/browse/ASTERISK-17264 Richard -- _ -- 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] Get second cipher in an extension
> how can I get the second character/cipher of an extension ?
>
> If I have : exten => 12345,n,NoOP()
>
> How can I get "2" ?
${EXTEN:1:1}
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridged Digital call
> > > Just upgraded from 1.6? to 1.8.4.1 > > > > > > > > > I ised to be able to get a digital call working across a bridged > > > isdn > > > channel in 1.6 and 1.4 using the following;- [snip] > > Could be a problem in the media stream handling not being setup for > > digital mode. > > > ..., should I report a bug on the issue tracker? Did anything change outside of Asterisk? (Different ISDN equipment or configuration for instance.) If not then yes I think it is a bug since you say it used to work with v1.4 and v1.6.x. I think it could be a problem in the media stream handling not being setup for digital mode. For completeness, the bug report should have attached: 1) chan_dahdi.conf (and any files it includes) 2) Debug capture files of "pri set debug on span x" output of a call attempt for the incoming call leg and the outgoing call leg. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridged Digital call
> > > Hi All
> > >
> > > Just upgraded from 1.6? to 1.8.4.1
> > >
> > >
> > > I ised to be able to get a digital call working across a bridged
> > > isdn
> > > channel in 1.6 and 1.4 using the following;-
> > >
> > >
> > > exten => _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
> > > exten => _X.,2,dial(DAHDI/g1/${EXTEN})
> > > exten => _X.,3,Noop(${CHANNEL})
> > > exten => _X.,4,hangup
> > > exten => _X.,5,(Set(CHANNEL(transfercapability)=DIGITAL)
> > > exten => _X.,6,dial(DAHDI/g1/${EXTEN})
> > > exten => _X.,7,hangup
> > >
> > >
> > > this still dials and aswers in 1.8 but no frames are passed and the
> > > call times out and drops
> > >
> > > I have also tried
> > >
> > > exten => _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
> > > exten => _X.,2,dial(DAHDI/g1/${EXTEN})
> > > exten => _X.,3,Noop(${CHANNEL})
> > > exten => _X.,4,hangup
> > > exten => _X.,5,Noop
> > > exten => _X.,6,dial(DAHDI/g1d/${EXTEN})
> > > exten => _X.,7,hangup
> > >
> > > with exactly the same outcome,
> >
> > Both of these methods should work after doing a quick look a the code.
> >
> > Does the outgoing call SETUP indicate digital capability?
>
> both show transfercapability DIGITAL
Could be a problem in the media stream handling not being setup for digital
mode.
Richard
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridged Digital call
> Hi All
>
> Just upgraded from 1.6? to 1.8.4.1
>
>
> I ised to be able to get a digital call working across a bridged isdn
> channel in 1.6 and 1.4 using the following;-
>
>
> exten => _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
> exten => _X.,2,dial(DAHDI/g1/${EXTEN})
> exten => _X.,3,Noop(${CHANNEL})
> exten => _X.,4,hangup
> exten => _X.,5,(Set(CHANNEL(transfercapability)=DIGITAL)
> exten => _X.,6,dial(DAHDI/g1/${EXTEN})
> exten => _X.,7,hangup
>
>
> this still dials and aswers in 1.8 but no frames are passed and the
> call times out and drops
>
> I have also tried
>
> exten => _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
> exten => _X.,2,dial(DAHDI/g1/${EXTEN})
> exten => _X.,3,Noop(${CHANNEL})
> exten => _X.,4,hangup
> exten => _X.,5,Noop
> exten => _X.,6,dial(DAHDI/g1d/${EXTEN})
> exten => _X.,7,hangup
>
> with exactly the same outcome,
Both of these methods should work after doing a quick look a the code.
Does the outgoing call SETUP indicate digital capability?
Richard
--
_
-- 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] CDRs in 1.8
> I'm using ISDN30 for a bridged application > > in all the old versions of asterisk the time slot number is shown in > the channels and dstchannel fields of the cdr > > I understand this has chaned in 1.8,is there a way of getting the time > slot information stored somewhere at the end of the call so this can > be interigated? Check the ChangeLog of your release to see if the fix to add CHANNEL(dahdi_channel) is present. The fix also added a new AMI DAHDIChannel event. Richard -- _ -- 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] sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway!
> Is it possible that the installation libpri-1.4.11.5 newer than the > libpri-1.4.11.5-patch? > > Well, when I typed (note: I am trying to apply the > libpri-1.4.11.5-patch for the libpri-1.4.11.5): > > libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch > > It gave me that patched detected as shown below (example of one file, > and I got same for other files): > > patching file pri.c > Reversed (or previously applied) patch detected! Assume -R? > > And when I reached to the q931, then it failed to save and apply the > hunk !! I don't know where you got that patch, but I would just use the libpri SVN brahches/1.4 code. It has the latest patches and fixes even after the libpri 1.4.12-beta3 tag. The committed "No D-channels available" fix only suppresses repetitions of that message (about every 4 seconds) it does not fix the cause of any communication problem. Asterisk and libpri have independently fixed this excessive repetition. The commits were done to the respective branches around November 2010. One of the newer fixes in the libpri SVN code that may apply to you involves the retransmission of I-frames. 1) Does "pri set debug 2 span 1" show message exchange with the PSTN or is it one way traffic? 2) Does the D channel bounce up and down or is it continuously down? 3) Does chan_dahdi complain of CRC errors? Richard -- _ -- 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] How asterisk use pri channel
> Extracted from chan_dahdi.c: > > Dial(DAHDI/pseudo[/extension[/options]]) > Dial(DAHDI/[c|r|d][/extension[/options]]) > Dial(DAHDI/![c|r|d][/extension[/options]]) > Dial(DAHDI/i[/extension[/options]]) > Dial(DAHDI/[i-](g|G|r|R)[c|r|d][/extension[/options]]) > > i - ISDN span channel restriction. > Used by CC to ensure that the CC recall goes out the same span. > Also to make ISDN channel names dialable when the sequence number > is stripped off. (Used by DTMF attended transfer feature.) > > g - channel group allocation search forward > G - channel group allocation search backward > r - channel group allocation round robin search forward > R - channel group allocation round robin search backward > > c - Wait for DTMF digit to confirm answer > r - Set distintive ring cadance number > d - Force bearer capability for ISDN/SS7 call to digital. > > > Where this information can be read, for those of us that don't read C > source code ? > Is there "foo show bar" command for that ? It should be in some dial help but it is not. The syntax is channel driver specific and every channel driver has its own needs. I created that comment block in chan_dahdi.c since I could not find it elsewhere when studying the code. Richard -- _ -- 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] How asterisk use pri channel
> But believe me no one option works for me. I tried dahdi/25/XXX > but it still using pri first channel or anyother channel Dialing DAHDI/25/ will dial channel 25 only. For ISDN spans I do not recommend doing that because your call will fail if that channel is already in use when there may be other B channels available for the call. > In old zap school you can do that but in dahdi I don't think you can. > Until unless you create g1 g2 ... Group in chan_dahdi.cfg and map > channels there. You should be creating groups for your ISDN spans. Richard -- _ -- 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] How asterisk use pri channel
> We have two pri line and I want to see how asterisk distribute > outgoing call per channels > > I meant it use first last channel 47 or it will use first channel? > > Or it will allocate dynamically ? Extracted from chan_dahdi.c: Dial(DAHDI/pseudo[/extension[/options]]) Dial(DAHDI/[c|r|d][/extension[/options]]) Dial(DAHDI/![c|r|d][/extension[/options]]) Dial(DAHDI/i[/extension[/options]]) Dial(DAHDI/[i-](g|G|r|R)[c|r|d][/extension[/options]]) i - ISDN span channel restriction. Used by CC to ensure that the CC recall goes out the same span. Also to make ISDN channel names dialable when the sequence number is stripped off. (Used by DTMF attended transfer feature.) g - channel group allocation search forward G - channel group allocation search backward r - channel group allocation round robin search forward R - channel group allocation round robin search backward c - Wait for DTMF digit to confirm answer r - Set distintive ring cadance number d - Force bearer capability for ISDN/SS7 call to digital. All are valid for v1.8 and trunk. The i option and ! option are not valid earlier than v1.8. Richard -- _ -- 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] Free CNAM
> The system uses real Telco CNAM Dips. Any generic names you get are > from the subscriber's carrier itself. We can only provide what we > ourselves get. There's more than one CNAM database (aren't there seven?). I would have hoped that a service such as this would look at a bunch of them and choose the one that had the best result. -- _ -- 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] Free CNAM
> Try them all again. Remember that this is a static database that has to > 'research' numbers it has not seen before. What happens when the CNAM is changed? How often does it go back and poll the database? -- _ -- 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] Free CNAM
> Try them all again. Remember that this is a static database that has to > 'research' numbers it has not seen before. Well, that doesn't make it very interesting: most calls I'd expect to get won't have been seen by it before. > By now (a few minutes later), the database should have been updated. I tried it, but it returns the same kind of junk that some of the databases do. For example, on a Florida number, it just says "FLORIDA" instead of the proper name (some of the CNAM databases have the right name). -- _ -- 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] Free CNAM
> FreeCNAM.org is providing a free CNAM API for Open Source PBX users. > This API queries a private CNAM database, and returns standard > 15-Character CNAM results. Any entry not already in the database will > be queued for investigation, and added to the database as soon as > information is located. This system has access to several CNAM > backends, and is not a party to any use-limiting or no-caching > agreements. > > The API is: http://freecnam.org/dip?q=2024561414 I just tried this on about a dozen numbers I have in various parts of the US (cell, business, and a landline number I've had for decades) and NONE of them were listed in this database. Indeed I can't find one that IS. -- _ -- 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] receive faxes
> I don't believe you really understand what Open Source means...it > does not mean "FREE". Actually, it DOES mean "free", especially since Asterisk is under the GPL. But, as RMS often says, "that's 'free' as in 'free speech', not 'free beer'". That problem doesn't exist in French, where there are two distinct words for "free" and they refer to it as "libre". The point of Free (or Open Source) Software is that if you think that some company (e.g., Digium) is charging too much to maintain it, you have the freedom to maintain it yourself of even start your own company and compete with them. -- _ -- 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] Issue with Asterisk & Aastra 57i at v3.2
> In that case it suggests it is some setting you have applied to the > phones that is causing it. I just called Aastra tech support. I'm always VERY impressed that the first person who picks up the phone is very technical. He said that they've had reports of this issue. The problem goes away when "sip rport: 1" is removed. Even though the manual says that it's "recommended" for NAT, it isn't needed and is causing this problem. They appear to be treating this as a bug. (There's supposed to be a new version of the firmware very soon, since it's ready, but just not on the web, but it won't solve this issue.) -- _ -- 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] Issue with Asterisk & Aastra 57i at v3.2
> In that case it suggests it is some setting you have applied to the > phones that is causing it. Can you post the local.cfg & server.cfg > files from the phone (removing the passwords from there first)? Sure: local.cfg is checksums, server information, and: contrast level: 3 ringer volume: 8 contact rcs: 0 sip line1 dnd: 1 Server.cfg removing server/account/password/button/softkey information is below. Note that I added the contact matching config when I ran into this problem, so it failed with and without it. # options simple menu: 1 time server disabled: 0 time server1: pool.ntp.org time server2: time.windows.com download protocol: HTTP web interface enabled: 1 auto resync mode: 3 auto resync time: 03:20 emergency dial plan: 911 sip dial plan: "[12367]xx|5xxx|*xxx|#[1-9]|9;1xx|9;[2-9]x|9;011x+#|8;x+#|#0[12367]xx|#05xxx|#0*xxx|#0#[1-9]|#09;1xx|#09;[2-9]x|#09;011x+#|#08;x+#x" sip proxy ip: gnat.com sip contact matching: 2 sip rport: 1 sip update callerid: 1 sip blf subscription period: 120 sip explicit mwi subscription period: 120 sip accept out of order requests: 1 sip use basic codecs: 1 sip silence suppression: 0 sip vmail: "#6" live dialpad: 1 directory 1: main.csv directory 2: special.csv dnd key mode: 1 call forward key mode: 1 call hold reminder: 1 call hold reminder timer:60 call hold reminder frequency:60 call waiting:1 call waiting tone period:5 bl on time: 600 https validate certificates: 0 xml get timeout 10 preferred line: 1 softkey selection list: none, speeddial, blf, dnd, callforward sip xml notify event: 1 time zone name: US-Eastern > But it might allow your users to make phone calls while you fix the > issue properly. Calls can be MADE with the new firmware, just not recieved. A better workaround for me is to just stay with the old firmware. -- _ -- 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] Issue with Asterisk & Aastra 57i at v3.2
> Asterisk does indeed send an Options before the OK but my 57i doesn't > seem to mind. That's odd. It does for me. > Perhaps you need to upgrade firmware on the Aastra phone? The problem occured when I DID upgrade it! Precisely to the one you mentioned. > Or turning off qualify for this peer might work-around it for you. I'm sure it would, but all peers are those phones, so that's not an acceptable workaround. -- _ -- 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] Issue with Asterisk & Aastra 57i at v3.2
> Is asterisk replying differently when firmware 3.2 is used ? No, but the phone cares with 3.2 and not with 2.6. -- _ -- 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] Issue with Asterisk & Aastra 57i at v3.2
I recently tried to update my Aastra 57i to version 3.2 and ran into a problem. It won't properly register and says "contact mismatch". I added "sip contact matching: 2" to aastra.cfg, but that didn't help. When I look at the SIP trace, but I see is the Aastra sending a REGISTER and Asterisk replying with the 401. The phone then sends the REGISTER again, this time with the hash. Asterisk now replies OK, but sends an OPTION packet FIRST and I think that confuses the Aastra. Has anybody seen this? Is there any way to have the packets sent in the proper order? -- _ -- 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] Odd error in libpri
> > Please create a mantis issue describing this problem. > > Pardon my ignorance, but what does "mantis" refer to? > Mantis is the issue tracker at: https://issues.asterisk.org Richard -- _ -- 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] Odd error in libpri
> Please create a mantis issue describing this problem. Pardon my ignorance, but what does "mantis" refer to? -- _ -- 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] Odd error in libpri
> As I'm reading this, libpri thinks that the SV8300 is complaining that > a "mandatory" IE is missing, in this case time/date. However, the > field is > THERE. But when I go back to a working libpri (r1878), I see that the > time/date is NOT sent on the CONNECT. > > If I'm reading Q.931 correctly, 5.1.8 (page 118) says that the > Date/time > IE may be included "as a network option". > > I see this was added to libpri at revision 2187, in response to issue > number 18047. > > I played around a bit. Since the spec includes seconds, I added > seconds > to see if that made it work, but it didn't. > > I DID work when I deleted Q931_IE_TIME_DATE from connect_net_ies. > > Whether or not it's a bug for the SV8300 to reject that IE, it's > likely > that NEC won't fix it. > > This likely means that a new config option is needed, but I think that > means it'd also have to be done in chan_dahdi.c in Asterisk in > addition > to libpri. Is that right? Yes chan_dahdi.c/sig_pri.c and libpri need to be modified to add a config option. Please create a mantis issue describing this problem. Thanks Richard -- _ -- 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] Odd error in libpri
I just updated libpri 1.4 on my system to the latest from that branch and my QSIG connection to an NEC SV8300 stopped working. The trace showing the problem is below: q931.c:5640 q931_connect: Call 7168 enters state 10 (Active). Hold state: Idle > DL-DATA request > Protocol Discriminator: Q.931 (8) len=21 > TEI=0 Call Ref: len= 2 (reference 7168/0x1C00) (Sent to originator) > Message Type: CONNECT (7) TEI=0 Transmitting N(S)=29, window is open V(A)=29 K=7 > Protocol Discriminator: Q.931 (8) len=21 > TEI=0 Call Ref: len= 2 (reference 7168/0x1C00) (Sent to originator) > Message Type: CONNECT (7) > [18 03 a9 83 81] > Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 > Exclusive Dchan: 0 > ChanSel: As indicated in following octets > Ext: 1 Coding: 0 Number Specified Channel Type: 3 > Ext: 1 Channel: 1 Type: NET] > [1e 02 81 82] > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 > Location: Private network serving the local user (1) > Ext: 1 Progress Description: Called equipment > is non-ISDN. (2) ] > [29 05 0b 05 01 0e 03] > Time Date (len= 7) [ 11-05-01 14:03 ] < Protocol Discriminator: Q.931 (8) len=13 < TEI=0 Call Ref: len= 2 (reference 7168/0x1C00) (Sent from originator) < Message Type: STATUS (125) < [08 03 81 e0 29] < Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) < Ext: 1 Cause: Mandatory information element is missing (96), class = Protocol Error (e.g. unknown message) (6) ] < Cause data 1: 29 (41) < [14 01 04] < Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Delivered (4) Received message for call 0x2aaab81d15c0 on link 0x1b0db440 TEI/SAPI 0/0 -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) As I'm reading this, libpri thinks that the SV8300 is complaining that a "mandatory" IE is missing, in this case time/date. However, the field is THERE. But when I go back to a working libpri (r1878), I see that the time/date is NOT sent on the CONNECT. If I'm reading Q.931 correctly, 5.1.8 (page 118) says that the Date/time IE may be included "as a network option". I see this was added to libpri at revision 2187, in response to issue number 18047. I played around a bit. Since the spec includes seconds, I added seconds to see if that made it work, but it didn't. I DID work when I deleted Q931_IE_TIME_DATE from connect_net_ies. Whether or not it's a bug for the SV8300 to reject that IE, it's likely that NEC won't fix it. This likely means that a new config option is needed, but I think that means it'd also have to be done in chan_dahdi.c in Asterisk in addition to libpri. Is that right? -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
> > Now imagine that 1.4 stays at only security level. For first case we > > have 2 options: upgrading for security reasons to last version but > > then no more voicemail, or staying with 1.4.26. In the second case, > > upgrading both servers to test with 1.8. If it's still not working, it > was time > > loose beside other problems. > > If there are obvious regressions in major functionality such as voicemail, > I'm more than happy to still consider making fixes for those problems during > the "security maintenance" period. It has to be pretty clear, though, and > in this particular case, it is. > > Voicemail has been through several issues. Can't remember the details, we experienced issues when imap was added. It broke the file based voicemails even when imap was not used. As long as major bugs, like this and deadlocks are taken care of during the 'security maintenance' period, most people are happy. New features should be only added as a separate patch for risk takers. The main branch should be major bugs only. -- _ -- 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] The new ConfBridge application is now in Asterisk Trunk!
> No, conference scheduling is not a feature that we have built > directly into ConfBridge, and I'm debating on what it would look > like. Scheduling isn't built into MeetMe either, but the fact that it dynamically reads from a database means that you can write external programs (such as Web-Meetme) that create conferences that MeetMe can read. For me, in order for ConfBridge to be at all interesting, it needs the same functionality. -- _ -- 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] The new ConfBridge application is now in Asterisk Trunk!
> To help get you started, Malcolm Davenport has written some fantastic > documentation on the asterisk.org wiki. It can be found below. I've looked at this documentation, but can't find the documentation the realtime interface, which is needed in order to schedule conferences in the future. Is the documentation missing or is this not part of the application? We depend on it HEAVILY. -- _ -- 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] Weird PRI error on an QUAD E1 span: Ring requested on unconfigured channel 255/255
If the call volume is low enough and the error reproducable enough, you may want to look at "pri set span 3 debug". -- _ -- 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] [1.4] Reading phone number the French way?
> exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,10,,2,20)
>
> For instance, a landline number in Paris like 01 42 92 81 00 is read
> "zero-one, forty-two, ninety-two, eighty-one, zero-zero", where I
> assume Americans would read all the digits individually ("zero, one,
> four, two, etc.")
Maybe something like:
exten => s,n,SayDigits(${NBR2CALL:0:1})
exten => s,n,SayNumber(${NBR2CALL:2:2})
exten => s,n,SayNumber(${NBR2CALL:4:2})
exten => s,n,SayNumber(${NBR2CALL:6:2})
exten => s,n,SayNumber(${NBR2CALL:8:2})
Or make changes in say.conf.
--
_
-- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
> I recognize all the options given yet as I explained before they are not > viable. I do not have the resources to pay someone, I do not have the > expertise to fix this issue because according to an asterisk developer > "any fix in that area would be deeply architectural in nature"... what > other options are there? In a commercial product, you have two options when you find a bug: (1) Pay for it to be fixed. (2) Live with the bug. In an open-sourced product, you have those same two options, plus an additional one: (3) Fix it yourself. Those are the only three you have to choose from. -- _ -- 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] Hide the plain text password
> > - The config file reader looks for strings of the form "{enc:}:
> > and replaces them, before otherwise parsing the line, with the decrypted
> > version of the string using the key in the "master_key" file.
>
> This sounds pretty reasonable, except perhaps that you might only want
> to convert strings in password fields -- otherwise you risk false
> positives in e.g. the dial plan.
I think this works much better if it's purely lexical. Otherwise, you
have to teach the code what's a password and what's not and maintaning
that is an ongoing issue, so I think a cleaner design would be to pick
some string that's just not going to occur anywhere.
> I can recommend contracting with one of the indepedent Asterisk
> developers to get this done. You will likely find them on the
> Asterisk-biz-list.
I could easily do it myself if it were something that I personally needed
(except that I'm not sure if two-way encryption routines already exist
in Asterisk), but we don't have enough passwords for this to be an issue.
I was posting the design to address the issues raised by the person who
started the thread.
--
_
-- 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] further action after caller in a queue hangs up
Tried that, not possible to play a sound file and prompt users to enter a number. Is there a way to revive the channel? On Tue, Feb 15, 2011 at 2:38 AM, Pezhman Lali wrote: > you can run any function in your hangup extension, > > exten => h,1,... > > best > On Tue, Feb 15, 2011 at 12:21 PM, Richard Zheng wrote: > >> Hi, >> >> In ACD queue, is it possible for the agent to take some actions when the >> caller hangs up? For example, to let the agent to enter some numbers for >> accounting purpose. >> >> 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 >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hide the plain text password
> #include the password (a file the line 'secret=') from a local file on
> the file system. The user has no access to it, right?
Right, but we're not talking ONE password, but ANY password. Having
dozens of those files, one for each password, gets to be a real pain
really fast. And you STILL want CM control of password changes even
if you're storing the encrypted versions: you want to be able to go
back to an old password, even if you don't know what it is.
> One test for you to consider: are the users able to use the "encrypted"
> configuration item in a different Asterisk system (without your
> concent)?
Of course not! It would be useless if that were the case: the whole
point here would be that you need the master encryption key.
Here's a possible design:
- There's optionally a file in the config
directory called "master_key". It contains just a string.
- A CLI command "core encrypt " is added to Asterisk. It takes the
provided string, encrypts it using the string in master_key, and outputs
a string of the form "{enc:}:
and replaces them, before otherwise parsing the line, with the decrypted
version of the string using the key in the "master_key" file.
--
_
-- 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] Hide the plain text password
> Right. But it really won't help much (except complicating things) if the > user has decent access to Asterisk. Yes, but we're talking about cases where the "user" *doesn't* have access to Asterisk. At many locations, including mine, Asterisk runs on a machine dedicated for that purpose and only people administering it have access to that machine. But config files are placed in a CM system which MANY more people have access to. Having plaintext passwords in those files is a real problem. -- _ -- 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] Hide the plain text password
> How does that improve things? The reason that works with Cisco routers > is because the code that reads that special key file and uses it to > decrypt the other files is closed-source; nobody can see how it works. > > As another poster said, that's not true for Asterisk. If Asterisk had > such a facility, the method used to decrypt the protected passwords > would be publicly available, as would the decryption key (in the special > key file). Anyone who wanted to decrypt the passwords from the config > files would have an only slightly more complex route to do so... it > would still be straightforward. Please reread what I wrote. The encryption key for the passwords wouldn't be in Asterisk sources, but selected BY THE USER and stored in a SINGLE configuration file that contains just that password. This is what Cisco does. That way, the rest of the config files, which you might want to put in a CM system, need not be protected. -- _ -- 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] Hide the plain text password
> Anyway, the answer is: No, it's mathematically impossible to do > that. Even if the passwords were stored encrypted, Asterisk itself > has to be able to get the plaintext passwords to send to the remote > server; so the code to decrypt them must necessarily be located on > the machine. And the Source Code to Asterisk is readily available, > which is how come you were able to benefit from it, so it would be > trivial to extract the passwords in any case. But there IS a way to improve things, and it's what Cisco routers do. You can have all password stored in config file encrypted with a single master key. That key is stored in a special file, containing just that key. THAT file must then be heavily-protected, but all OTHER config files can now be placed into CM or anywhere else they might be needed. -- _ -- 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] further action after caller in a queue hangs up
Hi, In ACD queue, is it possible for the agent to take some actions when the caller hangs up? For example, to let the agent to enter some numbers for accounting purpose. 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hide the plain text password
> Who are you hiding them from? Anyone with access to the Asterisk server > can already do far more damage than extracting these passwords. You may (like we do) want to store config files in a version control system in a common repository. People who have access to that repository don't necessary have access to the Asterisk server. -- _ -- 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] extconfig, realtime, and SIP
I'm confused about a few things relating to realtime, SIP and config in general. As I understand it, with the exception of extensions.conf, I can either have a config file completely in text or completely in a database. Is that correct? I can't find documentation for exactly what "switch =>" does but is that only in the dialplan and a way to have it partly from a file and partly from a database? For SIP, do I understand it correctly that I can have sip tables both via realtime AND in sip.conf? For the LDAP realtime, how can I implement "setvar"? -- _ -- 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] Top Posting
> It is not a matter of preference, it is actually a rule [1]. Top-posting > is also an annoying practice [2] and NOT the general accepted way to reply. And that's been the case for at least TWO DECADES. I find it amazing that this is still being argued now. -- _ -- 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] sip.conf, realtime, and LDAP
I'm confused exactly what's supported with LDAP and Asterisk. What I want to do is to have SIP peer information read directly (in realtime) from LDAP. Can this be done? If so, with what Asterisk versions? -- _ -- 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] + on Caller-ID
> What is the proper way to format a caller-ID here in the U.S.? > > Is it: > +15705551212 That's the correct one. > I've always seen it +15705551212, but as I understand it the country > code for the US is 011, which to me would indicate you put > 011-570-555-1212 as the callback number. The country code for the US is 1, which is why +1570... is correct. "+" means "this is an international call" and tells a cellphone, for example, to replace "+" with whatever is the international dialing code for the location where it's currently located. (In the US, that 011, so you'd dial 011-1-570-555-1212, which is the correct way to dial that number.) -- _ -- 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] Volume on meetme recording
It's kind of low for me. How does one control that volume? -- _ -- 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] Go from *100* to just 100
> how can I go from *100* to 100 ?
>
> I know I can do something like ${EXTEN:1} but that way I only get rid of just
> one *.
${EXTEN:1:-1} removes the first and last characters of ${EXTEN}.
--
_
-- 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] Res: digits in chan_dahdi
> I tried relaxdtmf = yes but has not worked. > > If I type very slowly digits are recognized normally. Then indeed it won't make a difference. If that were your problem, it likely wouldn't work at any speed. -- _ -- 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] digits in chan_dahdi
> I dial 12345678, but only '16 'is received by the asterisk. You may want to try relaxdtmf=yes in chan_dahdi.conf. That fixed a similar problem for me. -- _ -- 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] Error loading skype_for_asterisk
This suddenly started appearing and I'm not sure why. Any ideas? asterisk*CLI> module load chan_skype.so Unable to load module chan_skype.so Command 'module load chan_skype.so' failed. [Sep 15 11:08:25] WARNING[12274]: loader.c:429 load_dynamic_module: Error loading module 'chan_skype.so': /usr/lib/asterisk/modules/chan_skype.so: undefined symbol: sfa_send_chat_message [Sep 15 11:08:25] WARNING[12274]: loader.c:797 load_resource: Module 'chan_skype.so' could not be loaded. -- _ -- 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] Moving from DSL to T1
I work in a small office and have fallen into the role of network support based on knowing enough about networking to be dangerous. Our office is moving from DSL to a T1. Were using Asterisk as our PBX and I'm looking for hints or resources that might help me make the transition as error free as possible. Are there well known gotchas that I shoud be aware of? Thanks in advance, Richard Stuppi rich...@stuppi.com 626-221-8010 -- _ -- 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] Losing first DTMF digit (with ASR)
> if you use SpeechBackground, DTMF is under ASR control (returned in > SPEECH_TEXT(0) ). It is returned in SPEED_TEXT(0), but it's still being done by Asterisk, not the ASR engine. > Anyway, your other test indicates that the DTMF press > used to stop the prompt is being "eaten" by the ASR or Asterisk. Asterisk, since it's now just Read. I've turned on LOG_DTMF and don't see it there either, so it's lower level, within DAHDI most likely. -- _ -- 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] Losing first DTMF digit (with ASR)
> Just for grins, do this command
> /bin/grep "num sent" /var/log/VestecASRE/Port-10500_2010-09-07.log
> This should show you all of the DTMF processed by the grammars today.
It doesn't show any. Isn't DTMF processed by Asterisk and not the ASR?
Anyway, I can now reproduce this in a simpler case: In my dialplan,
I have
exten => 316,1,Answer(100)
exten => 316,n,MSet(TIMEOUT(d)=1,TIMEOUT(r)=10)
exten => 316,n,Read(X,adacore/main,5)
exten => 316,n,SayDigits(${X})
exten => 316,n,Goto(1)
I linked a DID to this and call it from my cellphone (works from a landline).
About 70% of the time, it chops off the first digit. It does NOT do this
if I'm not playing a file. So this seems to be related to not stopping
the prompt or getting the prompt and the returned "voice" (DTMF) confused
in some way.
>
>
>
>
> --
> _
> -- 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] Losing first DTMF digit (with ASR)
> So it's a PRI/DAHDI connection. Yeah, but with switchtype=qsig, though that difference isn't likely relevant here. > Is SpeechBackground the first item in the context? No. There are plenty of others, starting with an Answer(200). Then a whole bunch of Speech* applications to load grammar and such along with some Sets. > Have you tried putting a playback in front of the SB command (to delay > processing a bit? No. But if I do that, I'll INCREASE the chance of losing DTMF since people may start dialing before the SpeechBackground (though, as I said, it isn't this problem since I've seen it when I know I've waited). Also, this is intermittent so I want to minimize the number of experiments since it'll be very hard to see if I'm actually improving something. > What if any information can you gleen from the > Vestec log (/var/log/VestecASRE/Port*)? None since DTMF is sent, not speech. It just shows Add, activate, delete, and close. I don't know that I've actually seen a log when the first DTMF digit is missed (too hard to reproduce), so I don't know if it's different. -- _ -- 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] Losing first DTMF digit (with ASR)
> Who is the carrier that the calls are flowing in from? It's a Paetec PRI into an NEC SV8300, then QSIG from there to 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] Losing first DTMF digit (with ASR)
> 1. Vestec, Lumenvox or other?
Vestec
> 2. How many digits of DTMF are you aiming for (using SPEECH_DTMF_MAXLEN?)
6
> 3. Are you presenting DTMF back (verbose ${SPEECH_TEXT(0)}) ?
Similar. There's a NoOp that display what was originally that value in
the log.
--
_
-- 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] Losing first DTMF digit (with ASR)
> Is the message played very long/short? I play a lot of my speechbackground > messages with beep in front (speechbackground(beep&foo)) so my user doesn't > start hitting DTMF until the message starts playing. It's about six seconds. I've seen the problem myself and I'm dialing the first DTMF digit around 1/2 second into the prompt. -- _ -- 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] Losing first DTMF digit (with ASR)
I'm having a wierd problem. Somewhere around 1-2% of the time, the first DTMF digit dialed gets dropped. This is occurring during a SpeechBackground application call. If the caller reenters the digits when given a second chance, all is OK. Any suggestions how to debug this intermittent problem? -- _ -- 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] Monitor asterisk
Hallo Keane, I truly have a nagios server, up and running 24/7 -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- 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] Monitor asterisk
Thanks Nasri, I don't want to only be able to use the CLI because I need the Helpdesk and application support Unit to be able to monitor, and they are not all the techy with CLI and stuff.. On Sun, Aug 8, 2010 at 5:00 AM, Nasir Iqbal wrote: > Hi > > following asterisk cli commands can help > > show channels, show uptime and show sysinfo > > here is an example > > asterisk -x "core show sysinfo" > > On Sun, Aug 8, 2010 at 12:25 AM, Richard Zulu wrote: > >> >> Hey guys, >> >> I have my asterisk box running without a gui. I now need to monitor usage, >> calls, traffic of voice calls on this asterisk server. I cannot now install >> a gui because the configs will be wiped out, how can i go about monitoring >> all the above? >> >> -- >> Richard Zulu >> Managing Director >> Time Information Company >> P.O Box 31842 >> Clock Tower >> Kampala, Uganda >> www.time.co.ug >> >> Mobile :+256752624006 >> Skype: zulu.richard >> >> >> -- >> _ >> -- 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 >> > > > > -- > Nasir Iqbal > > ICT Innovations > http://www.ictinnovations.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 > -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- 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] Monitor asterisk
Hey guys, I have my asterisk box running without a gui. I now need to monitor usage, calls, traffic of voice calls on this asterisk server. I cannot now install a gui because the configs will be wiped out, how can i go about monitoring all the above? -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- 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
