Re: [asterisk-users] R2 error Seize Timeout
Hello, I must admit that I have never set up an asterisk system with R2 signalling. But from the config files point of view, you stated TE_SIG_MODE in wanpipe1.conf as ccs which should be cas, right ? If this does not help, you need to connect an external E1 Monitor. Regards, Hans Am 08.03.22 um 06:41 schrieb Carlos Chavez: > Last month we switched a Panasonic pbx with a Freepbx 16 > appliance. We use a single E1 in MFC/R2 (Mexico) with Telmex as a > provider. This was connected for a couple of days for testing with no > problems before the client moved offices to a new location. In the new > location we are now having a problem every few days where we get the > following error: > > [2022-03-07 07:30:11] ERROR[3469][C-004c] chan_dahdi.c: Chan 10 - > Protocol error. Reason = Seize Timeout, R2 State = Seize Transmitted, MF > state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08 > [2022-03-07 07:30:44] ERROR[29573] chan_dahdi.c: Chan 10 - Protocol > error. Reason = Seize Timeout, R2 State = Clear Forward Transmitted, MF > state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08 > [2022-03-07 07:32:15] ERROR[3704][C-004e] chan_dahdi.c: Chan 10 - > Protocol error. Reason = Seize Timeout, R2 State = Seize Transmitted, MF > state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08 > [2022-03-07 07:32:52] ERROR[29573] chan_dahdi.c: Chan 10 - Protocol > error. Reason = Seize Timeout, R2 State = Clear Forward Transmitted, MF > state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08 > > When we see that error the E1 will no longer send or receive > calls. Our solution has been to stop and restart Asterisk and > Wanconfig/Dahdi to restore service. Since restarting solves it I am > wondering if the problem is on my side and not on the providers. So far > it happens once or twice a week. When we report this to the provider > they simply state that the problem is on our side (it is their default > position) unless we can provide evidence to the contrary. Any > recommendations on how to debug this? > > Here is wanpipe1.conf: > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 4 > PCIBUS = 8 > FE_MEDIA = E1 > FE_LCODE = HDB3 > FE_FRAME = NCRC4 > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > TE_SIG_MODE = CCS > TE_HIGHIMPEDANCE = NO > TE_RX_SLEVEL = 430 > HW_RJ45_PORT_MAP = DEFAULT > LBO = 120OH > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_SPAN = 1 > TDMV_DCHAN = 16 > TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS > Blue Alarm and keep line down > #wanpipemon -i w1g1 -c Ttx_ais_off to > disable AIS maintenance mode > #wanpipemon -i w1g1 -c Ttx_ais_on to > enable AIS maintenance mode > TDMV_HW_DTMF = NO # YES: receive dtmf events from hardware > TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events > from hardware > HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation > enabled with nlp (default) > # OCT_SPEECH: improves software > tone detection by disabling NLP (echo possible) > # OCT_NO_ECHO:disables echo > cancelation but allows VQE/tone functions. > HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of > incoming media (must have hwdtmf enabled) > HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the > line - could break fax > HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo > cancelation > HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software > tone detection (possible echo) > HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio > level to be maintained (-20 default) > HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio > level to be maintained (-20 default) > HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to > be applied to tx signal > HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to > be applied to tx signal > > [w1g1] > ACTIVE_CH = ALL > TDMV_HWEC = NO > MTU = 8 > > Here is system.conf > > span=1,1,0,CAS,HDB3 > cas=1-10,11-15,17-31:1101 > echocanceller=oslec,1-10,11-15,17-31 > loadzone=mx > defaultzone=mx > > Here is chan_dahdi.conf > > signalling=mfcr2 > mfcr2_variant=mx > mfcr2_get_ani_first=no > mfcr2_max_ani=10 > mfcr2_max_dnis=4 > mfcr2_category=national_priority_subscriber > mfcr2_call_files=no > mfcr2_mfback_timeout=-1 > mfcr2_metering_pulse_timeout=-1 > mfcr2_allow_collect_calls=yes > mfcr2_double_answer=no > mfcr2_immediate_accept=no >
Re: [asterisk-users] Playing a beep/noise during a call
Le 07.02.19 à 12:59, Rui Mota a écrit : > Hi. > I am using an appliance as SIP-ISDN gateway that has a (not so nice) feature > of hanging up a call if it detects more than 10 minutes of silence from > the originating side. > The problem is that the calls are always originated from a digital system > that, most of the times, just sends silence, for more than 10 minutes, only > sporadically has people talking, so this makes the calls to hangup > automatically after 10 minutes... > > I need to inject some noise/tone, periodically (like every 9 minutes), on the > originating call to avoid this behaviour. What is the best way to > achieve it? > > I've seen the Extenspy or mixmonitor commands, but i need to make it > periodic... > > Thank you in advance. > Rui > Hello, maybe you could use the approach described in https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PERIODIC_HOOK regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to install and configure Dahdi from Debian Stretch repo ?
Hello Tzafrir, Am 2017-03-21 um 11:23 schrieb Tzafrir Cohen: > On Tue, Mar 21, 2017 at 09:36:21AM +0100, Olivier wrote: > > I'm still having some questions: > > 1. I can't find any /etc/init.d/dahdi file in my newly built system so > "service dahdi status" (or systemctl status dahdi) fails with: > Unit dahdi.service could not be found. > Shall I worry about this ? > No. As I mentioned, that is meaningless. So it is no longer possible to make changes on dahdi parameters without rebooting the machine ? I think in the early days it was possible something like this: make a change in /etc/dahdi/system.conf asterisk -rx "module unload chan_dahdi.so" service dahdi stop service dahdi start asterisk -rx "module load chan_dahdi.so" Regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI timing settings
On 2014-08-19 23:56, Jeff LaCoursiere wrote: Hello, I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. The telco has been working with their switch manufacturer and took the output of pri show span 1 from me and came back with this: quote--- Please check your timers below. How did you determine your settings? * Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T201: 1000 T202: 2000 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000Our switch: Telcordia National ISDN 2: Range 10 - 90 seconds, default 90 seconds. * T312: 6000 T313: 4000 T316: -1Our switch: Telcordia National ISDN 2: Range 10 - 120 seconds, default 30 seconds. N316: 2 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 ---end quote--- Now I have no idea what T309 or T316 represent, but it seems odd that timers and counters would cause such an odd result... and the failure is immediate, not after some amount of timeout. Can anyone shed light on these settings and tell me if they are configurable? I don't think that this has something to do with the timers. I recommend to enable debug of your pri connection to get a trace to a sprint phone number. There you should get the disconnect cause of the call. Then your provider should be able to help you out. Maybe the reason of the problem is an incorrect number setting. Regards Hans -- _ -- 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] multicastRTp
On 2014-08-08 21:54, Jerry Geis wrote: On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis ge...@pagestation.com mailto:ge...@pagestation.com wrote: I am using a cyberdata sip paging adapter and with the Dial(MulticastRTP/basic/IP:port) and with tshark I see the RTP data, my device looks like its accepting the data and I hear a click for my relay on my device so it would seem its accepting the call, however - I hear no audio... If I call using the dial plan everything seems to work... Is there an issue with using call files ? Channel: MulticastRTP/basic/239.168.3.10:11000 http://239.168.3.10:11000 It all seems to work, I see multicast audio, the unit answers, I just get no audio or crappy audio... Is the codec not set right in that case from a call file? How do I set the codec for multicastrtp in a call file? might make sense that speak live the codec is already established but from a call file there is no codec Any thoughts or how do I set the codec in a call file for multicast to try it? Please check this link and see if this applies to you: http://www.voip-info.org/wiki/view/Asterisk+MulticastRTP+channels Regards Hans -- _ -- 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] High Availability with Asterisk
On 2014-03-07 17:31, Paul Belanger wrote: On Thu, Mar 6, 2014 at 3:33 PM, Markus unive...@truemetal.org wrote: Hi Thorolf, Am 06.03.2014 16:21, schrieb Thorolf Godawa: Using (para-)virtualization with Xen could be an other option, on systems with low load this works reliable, but what happens on systems with high load? Are there any issues known about problems with the realtime, packet loss etc. because it runs in a VM? hmm, all my Asterisk'es run in (KVM) VMs, no issues there. But how is this related to high availability? I think it's not. :) I think the way to go for high availability (and scalability) is Kamailio! In a redundant setup, running on 2 separate physical machines (maybe in a VM, doesn't matter). Then you make them failsafe using whatever tool(s) available. Then you can set up 1, 2, 10 or 100 Asterisk behind Kamailio and any of them could fail (but 1 :-) ) and you will still be online. Sorry, for the stupid question, but what happens if Kamailio fails ? Thanks. regards Hans -- _ -- 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 Licensing Revisited - I'm Sorry!
On 2014-02-28 14:04, Tahir Almas wrote: 1) We do not perform any transcoding whatsoever. All recordings, and voice mail are in G729, and allow=g729 for all peers and in sip.conf. Is there anything else we need to perform g729 passthrough. More importantly are we still liable? Given that most vendors support G729, why do some still require the need to transcode? As earlier referred following quote from their site DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm You have to pay royalty fee for using their algorithm and it does not matter whether you are trans-coding or not however there is no restriction to pay their royalty fee under testing / evaluation environment. Hmm, wouldn't that mean, that every single ISP needs to pay the fee for passing G.729 data through their network ? They really do not transcode. If you do not transcode, you do not use their algorithm. This is my opinion, but I 'm not a layer. Regards Hans -- _ -- 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 TON number
On 2013-09-25 09:22, Endri Stefani wrote: Hi Greeting to all you out there. I am new at asterisk, I have been working with PLMN platforms telecommunication for 5 years with NSN and Huawei. We have recently built an asterisk PBX with Trixbox and connected it to our MSS using Digium E1 cards(ISDN). Everything went smoothly as there are tons of information out there, except for the TON number. If you have worked in Telecommunication you will know the importance of TON flexibility. All the posts online suggested to update under Chan_dahdi.conf: pridialplan = international prilocaldialplan = international or other TON value ,restart the platform and then trixbox1*CLI dialplan reload I have already done this with no success. Are there other changes I have to make in order to change dialplan? If you change something in chan_dahdi.conf, 'dialplan reload' wont help. You need to reload chan_dahdi. I 'm not quite sure if 'module reload chan_dahdi' reloads the TON settings. But you should be save by: module unload chan_dahdi.so and module load chan_dahdi.so which completly unloads and loads chan_dahdi. Regards Hans -- _ -- 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] RTP port ranges
Maybe you should open 11955 on you fw as well. This could be the rtcp port. Regards Hans On 2013-09-13 11:49, Jonas Kellens wrote: Hello, and when I define 11500 - 11954 it should use a random port in this range. Where is it stated that you MUST use 1-2 ??? Someone else please ? Jonas. On 09/13/2013 11:46 AM, Andrew Colin wrote: Because normally it will use a random port between them On 9/13/2013 11:43 AM, Jonas Kellens wrote: On 09/13/2013 11:41 AM, Andrew Colin wrote: Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Jonas. -- _ -- 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] passing '302 moved temporarily' back to the SIP provider
Hello,
I 'm looking for a way to pass the '302 moved temporarily' received from the
SIP device
back to the SIP provider.
Here is the setup:
Some SIP phones are connected to an Asterisk System version 1.8.
External connection to the public network is also done via SIP to a VoIP
provider.
Phone A has a CFW all calls to a phone number in public network (Mobile Phone)
incoming call to Phone A, Phone A sends back a 302 response. This results in:
-- Got SIP response 302 Moved Temporarily back from 192.168.2.188:5072
-- Now forwarding SIP/public-00e7 to
'Local/00662943825@internal_extensions' (thanks to SIP/25-00e8)
[May 7 12:22:15] NOTICE[31241]: app_dial.c:859 do_forward: Not accepting call
completion offers from call-forward recipient
Local/00662943825@internal_extensions-f967;1
and a Dial via Local channel to the SIP provider is made.
However, two channels are consumed and bandwith is wasted.
Second approach:
I tried to catch the redirect in the call forward context which is defined
before the Dial statement:
and do the redirect with app_transfer.
e.g. snippet from macro-stdexten:
exten = s,n,Set(_FORWARD_CONTEXT=from_sip_forward)
exten = s,n,Dial(${ARG2},${ARG4},tTfwW)
[from_sip_forward]
include = internal_devices
exten = _X.,1,Verbose(1,${CALLERID(num)} tries call forward to ${EXTEN} for
device ${CALLERID(rdnis)})
exten = _X.,n,Transfer(${EXT_TRUNK}/${EXTEN})
exten = _X.,n,NoOp(Transfer STATUS: ${TRANSFERSTATUS})
However, this does not work,
Is there a way to send the 302 response to the VoIP provider ?
Thanks.
Hans
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Re: [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000
You did not show how the Nortel side is configured, especially LD 17 ADAN configuration. Regards Hans On 2013-05-03 11:27, Danilo Dionisi wrote: I'm sorry, the mail is automatically send :p However, I am for the Asterisk, there are other external consultants for Nortel ... according to you can be out a patch for Asterisk to send the facility of CPND??? Danilo Il giorno venerdì 3 maggio 2013, Danilo Dionisi dionisi.dan...@gmail.com mailto:dionisi.dan...@gmail.com ha scritto: Thanks Dale for your answer. I am a consultant for a project for Banca D'Italia and we have to give birth to 20 branches with 1,200 snom phones ... oh my god I can not record 1200 names on Nortel!!! : '( Il giorno giovedì 2 maggio 2013, Dale Noll dn...@wi.rr.com mailto:dn...@wi.rr.com ha scritto: I retired my Nortel switch a couple of years ago, but I do not believe I ever got Asterisk - Nortel to pass the CPND, just the number. If I remember correctly, I had to enter then names manually in Nortel (LD 95?) for display on the Nortel endpoints. On Tue, Apr 30, 2013 at 11:30 AM, Danilo Dionisi dionisi.dan...@gmail.com mailto:dionisi.dan...@gmail.com wrote: Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI --- Asterisk QSIG ---SIP Trunk--- Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send Nortel on the display of the name of the person i'm calling ... why? example: Snom phone = Danilo 1001 Nortel phone = Marco 2002 If I call from Nortel to Asterisk, I have the display of the Snom Marco 2002 and the display of Nortel Danilo 1001; If I call from Nortel to Asterisk, I have the display of the Snom Marco 2002 and the display of Nortel 1001 This is my / etc / asterisk / chan_dahdi.conf [channels] cc_offer_timer=20 ccbs_available_timer=4800 ccnr_available_timer=7200 cc_recall_timer=20 cc_agent_policy=native cc_monitor_policy=native pridialplan=private prilocaldialplan=private context=default 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 facilityenable=yes callerid=asreceived ;Sangoma A104 port 1 [slot:4 bus:17 span:1] wanpipe1 switchtype=qsig context=from_nortel group=0 echocancel=yes faxdetect=incoming signalling=pri_cpe channel =1-15,17-31 -- _ -- 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] asterisk 11 - No pri set debug off
On 2013-03-03 18:41, Olivier wrote:
hello,
In a machine I've got :
CLI pri set debug off
No such command 'pri set debug off' (type 'core show help pri set' for other
possible commands)
CLI core show help pri
pri intense debug span no description available
pri service disable channel Remove a channel from service
pri service enable channel Return a channel to service
pri set debug {on|off|hex|inte Enables PRI debugging on a span
pri set debug file Sends PRI debug output to the specified file
pri show channels Displays PRI channel information
pri show debug Displays current PRI debug settings
pri show spans Displays PRI span information
pri show span Displays PRI span information
pri show version Displays libpri version
My setup is :
asterisk 11.2.1
libpri 1.4.14
dahdi 2.6.1
So inline CLI tells pri set debug off exists but I can't run it.
Did I miss something ?
I think you have missed the span.
pri set debug off span x
Regards
Hans
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Re: [asterisk-users] Deleting OLD Voicemails
Please check out the scripts located in contrib/scripts Regards Hans On 2012-05-23 11:42, Danny Dias wrote: Hi, thanks for your answers... Can i delete like this: rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.* Is that ok? will this break something? A little doubt here, once the user hears the voicemail using the phone, the message is automatically moved to Old folder, is that right? Many thanks! 2012/5/23 Mehmet Avcioglu meh...@activecom.net mailto:meh...@activecom.net You can delete old files, it won't break anything. Also to prevent saving files in multiple formats, edit voicemail.conf and change format parameter under general. -- Mehmet Avcioglu meh...@activecom.net mailto:meh...@activecom.net On May 23, 2012, at 1:03 AM, Danny Dias wrote: Thanks Jason, But how to delete them? there are a lot of old voicemails, but i don't want to break the app_voicemail. 2012/5/22 Jason Parker jpar...@digium.com mailto:jpar...@digium.com On 05/22/2012 04:54 PM, Danny Dias wrote: There are 4 files for each voicemail: msg.gsm msg.txt msg.wav msg.WAV That is perfectly normal. The .txt file is metadata that contains things like caller ID and duration. Asterisk will also save voicemails into every format you have specified in voicemail.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
[asterisk-users] disable dahdi pri
Hello, is there a way to disable a span for maintenance purpose (i.e. send yellow alarm) ? What would be the correct ioctl definition ? DAHDI_MAINT seems not to be the right candidate. Would DAHDI_SHUTDOWN send an alarm ? Thanks Hans -- _ -- 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] Low cost BRI gateway
On 2012-03-13 18:38, Chris Bagnall wrote: Greetings list, I'm trying to source a very basic ISDN BRI - SIP gateway. Unfortunately, everything I've seen seems to want to do lots of other things - registering handsets, IVRs, voicemail, etc. I only want it to present an ISDN BRI as a SIP account - I have an asterisk server for the other stuff. :-) In any other environment I'd just use one of the Digium ISDN PCIe cards, but in this case the ISDN lines come into one building and the asterisk servers are in the other building across the road, and there's no copper link between them. Any suggestions gratefully received. http://www.switchvoice.com/ or http://www.teles.de/en/products-and-solutions/access-gateways/voip-gateways/voipbox-bri/ or http://www.patton.com/ Regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
Its the sending complete IE. I'm using EuroISDN and I also use overlap signalling on this interface. Regards Hans On 2012-01-10 01:12, C F wrote: Exactly which IE message are you trying to push manually? you shouldn't have to do that, it should be done in the configs for you. On Mon, Jan 9, 2012 at 1:45 PM, Johann Steinwendtner steinwendt...@gmx.net wrote: On 2012-01-09 17:46, Alex Villacís Lasso wrote: I am trying to collect information regarding a bug report for Elastix (http://bugs.elastix.org/view.php?id=1146). In this bug, an user has asterisk-1.8.7 and dahdi-2.4.1.2. He is trying to make an outbound call through an ISDN trunk, by placing Dial(DAHDI/g0/12345w) in order to send 12345, then wait a period, then send . I am still waiting for a response on what particular telephony card he uses, and the kind of ISDN setup (T1/E1/BRI) being used, but I want to know: Is this dialstring expected to work with an ISDN trunk? If so, are there any configurations that might cause it to stop working? The user claims that this same dialstring worked with Elastix 1.6 which had dahdi-2.2.0.2 and asterisk-1.4.26.1. Some additional information: the user reports that the dial attempt fails with hangup cause 28. From http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293 http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293 [^ http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293] : Code No. 28 - invalid number format (address incomplete). This cause indicates that the called party cannot be reached because the called party number is not in a valid format or is not complete. Is it actually possible that the code is trying to send a string of 'w's through the ISDN link? Or am I misunderstanding? I'm using 'w' to force sending the 'sending complete' IE in an ISDN setup message. But I don't know the usage of multiple 'w' in the dialstring. regards Hans -- _ -- 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] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
On 2012-01-09 17:46, Alex Villacís Lasso wrote: I am trying to collect information regarding a bug report for Elastix (http://bugs.elastix.org/view.php?id=1146). In this bug, an user has asterisk-1.8.7 and dahdi-2.4.1.2. He is trying to make an outbound call through an ISDN trunk, by placing Dial(DAHDI/g0/12345w) in order to send 12345, then wait a period, then send . I am still waiting for a response on what particular telephony card he uses, and the kind of ISDN setup (T1/E1/BRI) being used, but I want to know: Is this dialstring expected to work with an ISDN trunk? If so, are there any configurations that might cause it to stop working? The user claims that this same dialstring worked with Elastix 1.6 which had dahdi-2.2.0.2 and asterisk-1.4.26.1. Some additional information: the user reports that the dial attempt fails with hangup cause 28. From http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293 http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293 [^ http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293] : Code No. 28 - invalid number format (address incomplete). This cause indicates that the called party cannot be reached because the called party number is not in a valid format or is not complete. Is it actually possible that the code is trying to send a string of 'w's through the ISDN link? Or am I misunderstanding? I'm using 'w' to force sending the 'sending complete' IE in an ISDN setup message. But I don't know the usage of multiple 'w' in the dialstring. regards Hans -- _ -- 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] digium HW echocancellation - fax tone detection
Hello !
I 'm using a TE405P with a HW echocanceller module attached on it.
dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0.
As far as I know, the fax tone detection is done on the FW board.
How can I verify that the echo canceller has been turned off ?
When I do a cat /proc/dahdi/1 for span 1 I see still the VPM450 entry
near the channel although this was a fax call with CED tone.
1 TE4/0/1/1 Clear (In use) (EC: VPM450M)
Greping through the source, i see only a CED tone detection
entry in:
static const struct dahdi_echocan_features vpm450m_ec_features = {
.NLP_automatic = 1,
.CED_tx_detect = 1,
.CED_rx_detect = 1,
};
Does it mean that CNG tone is not going to be detected ?
Does the CED tone detection routine also detect ANSam tone for Super G3 fax ?
Thanks
Regards
Hans
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Re: [asterisk-users] UDPTL T38 via NAT
On 2010-06-22 12:36, Remco Bressers wrote: Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) This means my outgoing udptl traffic is correctly translated, but somehow i'm sending 172.16.0.156 instead of my public IP address on the firewall. Did you try t38pt_usertpsource=yes ? Best regards Hans -- _ -- 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] UDPTL T38 via NAT
On 2010-06-22 15:16, Remco Bressers wrote: On 06/22/2010 02:51 PM, Johann Steinwendtner wrote: On 2010-06-22 12:36, Remco Bressers wrote: Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) This means my outgoing udptl traffic is correctly translated, but somehow i'm sending 172.16.0.156 instead of my public IP address on the firewall. Did you try t38pt_usertpsource=yes ? Hi, Yes, i tried adding that to the SIP peer configuration for the FAX ATA. Should i put it on the PBX trunk configuration also?? Remco Yes. -- _ -- 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] Does Playback will answer the call?
Zhang Shukun wrote: hi, all in my test,it shows Playback will answer the call automaticly, but i don't want to so. i will use answer function to answer the call. could you help me ? core show application Playback Regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax, T38 and NAT
Magnus Benngård wrote: Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38 on 0197673581, changing t38pt_udptl=no to t38pt_udptl=yes,fec and try to send from 0197673581 to 0851711201, it is not working, switches to T38 sendimg a lot of UDPTL packages but it looks like (at least for me) that addresses are wrong. UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) 90.230.92.67 is WAN ip of 0197673581's router. 10.242.20.149 is ip of 0851711201's ATA (SPA2102). Shouldn't the UDPTL stream go through Asterisk? Have i missed sometheng else? Asterisk SVN-trunk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) 0197673581 disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) 0851711201 disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=yes context=inputinterior.se directmedia=yes dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=no qualify=yes sendrpid=pai t38pt_udptl=yes,fec transport=udp trustrpid=yes type=friend videosupport=no [0851711290] secret=xyz callerid=Input Interior Sundbyberg (fax) 0851711290 ... rest is the same as [0851711201] Regards, Magnus Maybe you should give t38pt_usertpsource=yes a try. Regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting a wait in a sip dial
Kevin P. Fleming wrote: David Gibbons wrote: snip This doesn't work? Dial(SIP/*31#ww061234123412) /snip When I was browsing the sip debugs, it seemed that the 'w' was not being honored for one reason or another. My thought at the time was maybe it didn't work at all over SIP. Does the w *just* work with dahdi or does it work over sip as well (assuming the provider honors it)? 'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). hmm, I use 'w' on ISDN channels (libpri) to signal sending complete, like Dial(DAHDI/g1/0123456w). But I did not know that 'w' means actually 'wait'. Regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P
Olivier schrieb: 2010/1/7 David Backeberg dbackeb...@gmail.com mailto:dbackeb...@gmail.com On Wed, Jan 6, 2010 at 6:23 PM, Olivier oza-4...@myamail.com mailto:oza-4...@myamail.com wrote: The second time I'm dialing an internal extension attached to the same ReceiveFAX application : 2. sendfax/hylafax/iaxmodem asterisk spandsp In the 2nd case, I've got 3 craches out of 3 attempts (with a rough estimee, the crash occurs 2 or 4s after ReceiveFAX's start). Before wasting anybody's time and effort within Asterisk support team, I would like to double check here if the case that crashes Asterisk is within specifications of involved apps. In other words, can you normally use Hylafax to send faxes to inner extensions or do you hace to stick to PSTN numbers ? I've never successfully done what you're trying to do, so I came to the conclusion that it was not supported. My reasoning is exactly the same as yours. It would be great if a developer could drop in and tell if this feature is supported or not. What asterisk version and spandsp version do you use ? IMO a crash can never be an answer if this kind of setup is supported or not. regards Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hard Phone with SMS
randulo schrieb: 2009/10/9 Juan E. Rodríguez jerdg...@gmail.com: Does any one know about a SIP hard phone capable of sending SMS messages (Or a SIP MESSAGE) that could be read from Asterisk dial plan?? The Gigaset S675IP series of DECT/SIP phone has SMS capability, but not sure it can work with Asteris. Yes, they do. (app_sms) Make sure you have installed the latest FW. Before, they sent the SMS out on the analog port only. Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
Mindaugas Kezys schrieb: Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified In the given example: *ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000* How do I interprete the jitter value ? Is the value 0.020917 good ? Bad ? Is there a unit behind this value ? Thanks Regards Hans *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA *Sent:* 2009 m. rugsėjo 22 d. 09:28 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell *ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000* if any one know plese help me to or give any documentation link regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iax2_read: I should never be called - issue 8286
Hello ! I 'm having a machine running asterisk 1.6.0.10 with IAX and dahdi. The calls are going in and out from IAX2 to dahdi (chan_dahdi + libpri) and vice versa. After a period of time, I got the following scenario: NOTICE[860] chan_iax2.c: I should never be called! WARNING[752] channel.c: Exception flag set on 'IAX2/iax-peer-13262', but no exception handler WARNING[752] channel.c: Exception flag set on 'IAX2/iax-peer-13262', but no exception handler WARNING[860] channel.c: Exception flag set on 'IAX2/iax-peer-13262', but no exception handler NOTICE[860] chan_iax2.c: I should never be called! NOTICE[752] chan_iax2.c: I should never be called! Those messages has been spit out for a few seconds till asterisk crashed. I found bugreport #8286 with a similar symptom. It notes that the patch on this issues has been integrated. But I could not find anything similar in chan_iax2.c nor in channel.c in 1.6. Has this fix been pulled out again ? Thanks regards Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] german voiceprompts
Hello ! Are there any plans at Digium to include also german voice prompts ? Thanks regards Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [hylafax-users] No Carrier detected sendig fax with Hylafax-Iaxmodem-Asterisk
Lee Howard schrieb: fred wrote: That’s being said, before going through the T38 Gateway tests, I’ve tried first the Fax2mail and Mail2fax solution with (Hylafax + Iaxmodem + Asterisk), to make a well-tested Asterisk solution working and I’m already facing some problems. Receiving faxes is ok but sending faxes gets stuck into “No Carrier Detected”. Debugging leads me to the following remarks: I) _Carrier or Called Station Identifier or (CED) tone_ from the called fax machine is received since recorded (iaxmdm-iax.wav attached) The audio coming from the receiver is quite audibly corrupted (just listen to it and then compare to when you call a fax machine over TDM). No doubt you are attempting to pass fax audio through VoIP. Please read: Is there a tool where I can feed the captured audio (e.g. extracted from tcpdump) in order to get a fax image ? Can hylafax in a certain mode do that ? Thanks. Regards Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] precision of wait dialplan application
Hello !
In order to chase after a problem I implemented the following dialplan to have
an
answertime of exactly one minute:
exten = xxx,1,NoOp(Test wait)
exten = xxx,n,Answer
exten = xxx,n,NoOp(Current timestamp:
${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)})
exten = xxx,n,Wait(60)
exten = xxx,n,NoOp(Current timestamp:
${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)})
exten = xxx,n,Hangup
But it seems the Wait(60) lasts longer than 60 seconds:
-- Executing [...@from_meridian:1] NoOp(DAHDI/29-1, Test wait)
in new stack
-- Executing [...@from_meridian:2] Answer(DAHDI/29-1, ) in new
stack
-- Executing [...@from_meridian:3] NoOp(DAHDI/29-1, Current
timestamp: 20090506135813) in new stack
-- Executing [...@from_meridian:4] Wait(DAHDI/29-1, 60) in new
stack
-- Executing [...@from_meridian:5] NoOp(DAHDI/29-1, Current
timestamp: 20090506135915) in new stack
-- Executing [...@from_meridian:6] Hangup(DAHDI/29-1, ) in new
stack
What is wrong in this example ?
Regards
Hans
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Re: [asterisk-users] Asterisk to Ericsson MD110 on E1 with ISDN-USR (not QSIG)?
Tony Mountifield wrote: I have been asked by a potential customer whether we can connect an Asterisk box to an Ericsson MD110 that has an E1 port with ISDN-USR. They are unable or unwilling to upgrade their E1 port to QSIG. Has anyone here had experience of successfully making such a connection? I have found a couple of hits on Google that suggest it should work, but I'm after something a little more definitive, based on actual experience, if possible. Can anyone tell me what the USR part of ISDN-USR actually means? Thanks Tony I would assume that USR is the user side of the connection, where the other side is the NET. The * term is CPE. But that does not describe which protocol they are using (EuroISDN, QSIG, etc..). But thats only a guess. Best regards Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
John Todd wrote:
Just a suggestion: have you tried more recent versions of Asterisk
with IAX2? I'm uncertain what version you're using, and if it's
1.2.4, that's getting to be quite old and the problems that you
reference may already be solved in more recent updates.
In addition, if you're set on SIP, there are features in newer
versions of Asterisk which allow you to both set and read SIP headers,
so you can insert values in those headers between Asterisk instances
which could then be used by the dialplan to split your calls apart
into different contexts or behaviors.
See function SIP_HEADER and application SIPAddHeader for the most
recent versions of Asterisk.
JT
On Mar 6, 2009, at 11:29 AM, tracinet wrote:
That stinks... We are migrating to SIP from IAX2 at the moment and
running into the same exact problem. No way to control the
destination context unless you use the fromuser. Of course that
is rendering Caller ID useless as you pointed out.
I am still researching this though, if I find anything I will post
it here...
On Fri, Mar 6, 2009 at 2:13 PM, Adam Robins
arob...@pharmacentra.com wrote:
no
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
] On Behalf Of tracinet
Sent: Friday, March 06, 2009 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using
SIP
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
arob...@pharmacentra.com wrote:
I am switching from IAX2 to SIP for my inter-Asterisk transport due to
assorted quality issues following the 1.2.4 upgrade.
On the server that SENDS the call, I have the following in SIP.CONF:
[192.168.1.2_OB]
type=peer
fromuser=OB
host=192.168.1.2
And in EXTENSIONS.CONF
exten = 91NXXNXX,1,Dial(SIP/${ext...@192.168.1.2_ob)
On the RECEIVING Server in SIP.CONF:
[OB]
type=user
context=longdistance
I am not using a REGISTER statement on the receiving server.
My problem is that the only way I can seem to get the call delivered
into the proper SIP context on the receiving box is to use the
fromuser=OB on the sending machine. I tried using username=OB,
but
then it delivers into the default context. I don't want to use
fromuser because it overrides the callerid.
You should be able to solve the callerid issue by using the sendrpid and
trustrpid prompts.
Hans
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[asterisk-users] colorized logfiles in asterisk 1.6.0.6
Hello ! I've upgraded our testsystem from asterisk 1.4.21 to asterisk 1.6.0.6. We 've noticed that the log files are now in colour. I could not find a note in the upgrade section about this. Is this a feature or a bug ? It might be usefull to have them not in colour. best regards Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colorized logfiles in asterisk 1.6.0.6
Danny Nicholas wrote: The log files themselves are not in color. It would be a style sheet change on the GUI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johann Steinwendtner Sent: Friday, March 06, 2009 2:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] colorized logfiles in asterisk 1.6.0.6 Hello ! I've upgraded our testsystem from asterisk 1.4.21 to asterisk 1.6.0.6. We 've noticed that the log files are now in colour. I could not find a note in the upgrade section about this. Is this a feature or a bug ? It might be usefull to have them not in colour. best regards Hans Sorry, that I wasn't clear enough. The logfiles contains escape codes + the colour codes. e.g.: [Feb 12 13:38:30] VERBOSE[19816] logger.c: == Registered custom function 'ESC[1;36;40mSQL_ESCESC[0;37;40m' [Feb 12 13:38:30] VERBOSE[19816] logger.c: func_odbc.so = (ESC[33;40mODBC lookupsESC[0;37;40m) [Feb 12 13:38:30] VERBOSE[19816] logger.c: == Registered application 'ESC[1;36;40mReadFileESC[0;37;40m' I do not use a GUI. Thanks. Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN Cause Code 100, Bosch Integral Management Connection
Wolfgang Pichler wrote:
Hi all,
we have the following setup
PSTN 3 PRI Lines --- Asterisk (1.4.22) --- Siemens HiCom
--- Bosch Integral
The Asterisk Machine does play the man in the middle - and adds some
extra functionality to the system (SIP users...) - the normal calls are
getting 1:1 through the system (incoming calls from PSTN are handled by
a simple Dial(ZAP/g1/${EXTEN}) (g1 = Siemens side) - so no special
handling here...
Everything is working as it should - beside of one little thing. The
Bosch Integral PBX does have a special extension (99) which is used to
remote manage the machine - this managment connection is working fine
without asterisk, as soon as asterisk is connected in the middle the
management connection wont work any more - getting back isdn cause code
100. I have already tried dial options d und c (make it digital - clear
channel) - no success.
Can you also post the incoming setup message to your asterisk system ?
They should be almost identical.
Best regards
Hans
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Re: [asterisk-users] ATA for large networks
Gordon Henderson wrote: On Mon, 29 Sep 2008, Andres wrote: In other words, I'd really appreciate feedback from voip administrators (not from resellers) who have had experience testing their devices and are happy with them. I would recommend the Linksys SPA8000 (8 port ATA). It is as solid and reliable as the SPA2102. The OP has 300 phones. That's 38 SPA devices. And while you might think it's solid and reliable, I have one customer using 3 of them and they're not impressed with echo on their existing analog network. This is high-end channel bank territory. Multiple E1s - traditional channel banks, or something like multiple 24-port Xorcom units or the like... Gordon Does each FXS port need one channel on the E-1 interface, or is there some sort of concentration ? I can imagine that in a hotel environment, 1 FXS port to one E1 channel is a waste of resources. Thanks Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan
Kevin P. Fleming wrote: Benoit Plessis wrote: Is it possible on a TE220p to deactivate the hardware echo canceler at will ? (With a function in the dialpan for example) example for fax SDA ,beeing able to shutdown the echo canceler could give better results don't you think ? All echo cancelers using Zaptel/DAHDI already disable themselves when FAX or modem communications are used, based on reception and detection of the CED tone that FAX machines and modems generate to make that happen. You can tell this happened by looking at the channel in Asterisk using 'zap show channel' or 'dahdi show channel' as it will show you that the echo canceler was disabled automatically. I thought the ec gets disabled only by the ec disable tone and not the CED tone. regards Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls on E1 TDMoE span are dropped at random
Florian Hackenberger wrote: On Tuesday 13 May 2008, Steve Totaro wrote: You can be shot several times and not die. I would try resetinterval=never just to be able to to say Not the problem rather than Probably not the problem. I'll do that, although I'm pretty sure that the setting is not the problem as the yellow alarm occured quite often a few minutes after restarting asterisk and the default is 3600 seconds. PRI debug info would be a great help too. The log I sent in the original message contains pri debug messages. I just had another look at it. I did not follow the thread, but can this be a timing problem ? It might be that the far end goes into maint mode due to slips, or what ever. Regards Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pridialplan/prilocaldialplan
Christoph Fürstaller schrieb: Can someone explain what the parameters pridialplan and prilocaldialplan are? What do they do and do I need them? I've connected an asterisk box via E1 (sangoma) to an alcatel 4200 pbx. The pbx technican complains about the format of the nr asterisk sends. Asterisk sends all numbers in on piece the pbx expects the numbers devided into international prefix, national prefix, phone number and extension. How can I set this behaviour? Is this possible with the above mentioned parameters? Or do I need something else/different? I hope someone can explain that to me. Hello ! pridialplan/prilocaldialplan sets the type of number information for the called/calling number. You need to be more specific which number you mean (called or calling) The pbx technican is wrong unless the alcatel uses a different protocol than EuroISDN or QSIG. Regards Hans ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T.38 faxing with spandsp and Grandstream HT.486
Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator inititate a re-invite ? txfax as originator: T.38 fax exchange takes place but the transmission is not successful, txfax reports errorcode 60 (Disconnected after permitted retry). Can someone recommend a T.38 able ATA which is working with spandsp ? Are there any terminals known which has been tested against spandsp ? Thanks ! Best regards Hans P.S.: Asterisk 1.2.7.1 with patched T.38 patch and patched app_rxfax.c app_txfax.c and udptl.c. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridging of PRI calls
Matthew Fredrickson schrieb: On Oct 12, 2006, at 1:17 PM, Johann Steinwendtner wrote: Hello ! I 've some questions how bridging of ISDN calls is done. Assume an asterisk system with a TE405 card equipped. (PRI1 - PRI4) An incoming ISDN call on PRI1 is transfered back to PRI3. Unless there is DTMF detection or other things involved, the bridging is done without Asterisk. Does this card have a some sort of cross connection ? Does the PCM leave the card ? Or is there some DMA magic in- volved ? Assume an asterisk system with two TE405 cards equipped. An incoming ISDN call is transfered back to the second TE405 card. Does this card have a seperate bus like H.100 ? How is the bridging done in this scenario ? If the call is between two spans on the card, there is an internal H.100-like bus that cross connects the timeslots. Matthew Fredrickson How is the situation between two cards ? Is there a kind of DMA mechanism involved, or does asterisk cross connect the timeslots ? Thanks ! Hans ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridging of PRI calls
Matthew Fredrickson schrieb: On Oct 12, 2006, at 1:17 PM, Johann Steinwendtner wrote: Hello ! I 've some questions how bridging of ISDN calls is done. Assume an asterisk system with a TE405 card equipped. (PRI1 - PRI4) An incoming ISDN call on PRI1 is transfered back to PRI3. Unless there is DTMF detection or other things involved, the bridging is done without Asterisk. Does this card have a some sort of cross connection ? Does the PCM leave the card ? Or is there some DMA magic in- volved ? Assume an asterisk system with two TE405 cards equipped. An incoming ISDN call is transfered back to the second TE405 card. Does this card have a seperate bus like H.100 ? How is the bridging done in this scenario ? If the call is between two spans on the card, there is an internal H.100-like bus that cross connects the timeslots. Matthew Fredrickson How is the situation between two cards ? Is there a kind of DMA mechanism involved, or does asterisk cross connect the timeslots ? Thanks ! Hans ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bridging of PRI calls
Hello ! I 've some questions how bridging of ISDN calls is done. Assume an asterisk system with a TE405 card equipped. (PRI1 - PRI4) An incoming ISDN call on PRI1 is transfered back to PRI3. Unless there is DTMF detection or other things involved, the bridging is done without Asterisk. Does this card have a some sort of cross connection ? Does the PCM leave the card ? Or is there some DMA magic in- volved ? Assume an asterisk system with two TE405 cards equipped. An incoming ISDN call is transfered back to the second TE405 card. Does this card have a seperate bus like H.100 ? How is the bridging done in this scenario ? I could not find useful information on the Product sheet, sorry. Best regards Hans ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI from Asterisk to Meridian
May be you can build an application which controls the background terminal of the Meridian. (This would be a serial connection to the M1) This application sends background commands like: se mw 3000. This could be a try. Best regards Hans Andrew Kohlsmith schrieb: Please keep responses to the list, so this can help everyone. On Tuesday 01 August 2006 09:26, you wrote: Thak you for you response. My interconection between Asterisk (Voicemail) and my meridian is througth PRI T1, so the only stuff that i can't activate is the light in the meridian digital phones, i understand the asterisk see those phones like a external devices, but i don't know is somebody create o modify the SIP MWI and generate TDM messages to meridian. This isn't about modifying Asterisk to work with the Meridian. This is about the Meridian simply having no way to accept that information from an external trunk. There are VM message centers but they are extraordinarily limited and you can't give a unique one to every user, or even to a group of users. They're line-based. Similarly, you can buy an expensive NAPN or MCDN license which will allow the Norstar to see a PRI as an internal trunk line, but now you are running an undocumented and proprietary PRI signaling protocol called SL-1. It's what Norstar systems use to communicate with each other (imagine two Norstar systems connected together over a leased T1). We have no documentation on it, and Nortel is very likely unwilling to give us the information. So, as I said, you are stuck using a Nortel ATA and an FXS port on Asterisk and using a hookflash *1 sequence to toggle it. Unfortunately the VM callback # will be the ATA's DN, so only one person at a time can access voicemail. I spent some time digging into this last year, but came up without an acceptable solution. I may be forgetting or misremembering some of the details but the end result is the same: you can hack something into it but it's a shitty solution. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI from Asterisk to Meridian
From voicemail.conf: ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail is left, delivered, or your voicemailbox ; is checked, uncomment this: ;externnotify=/usr/bin/myapp Maybe this approach can send the commands to the M1. Best regards Hans kritikus Araklidas schrieb: Yeah is true.but we have to sincronize this console command with Asterisk SIP MWI Regards. Cris. From: Johann Steinwendtner [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] MWI from Asterisk to Meridian Date: Tue, 01 Aug 2006 17:06:41 +0200 May be you can build an application which controls the background terminal of the Meridian. (This would be a serial connection to the M1) This application sends background commands like: se mw 3000. This could be a try. Best regards Hans Andrew Kohlsmith schrieb: Please keep responses to the list, so this can help everyone. On Tuesday 01 August 2006 09:26, you wrote: Thak you for you response. My interconection between Asterisk (Voicemail) and my meridian is througth PRI T1, so the only stuff that i can't activate is the light in the meridian digital phones, i understand the asterisk see those phones like a external devices, but i don't know is somebody create o modify the SIP MWI and generate TDM messages to meridian. This isn't about modifying Asterisk to work with the Meridian. This is about the Meridian simply having no way to accept that information from an external trunk. There are VM message centers but they are extraordinarily limited and you can't give a unique one to every user, or even to a group of users. They're line-based. Similarly, you can buy an expensive NAPN or MCDN license which will allow the Norstar to see a PRI as an internal trunk line, but now you are running an undocumented and proprietary PRI signaling protocol called SL-1. It's what Norstar systems use to communicate with each other (imagine two Norstar systems connected together over a leased T1). We have no documentation on it, and Nortel is very likely unwilling to give us the information. So, as I said, you are stuck using a Nortel ATA and an FXS port on Asterisk and using a hookflash *1 sequence to toggle it. Unfortunately the VM callback # will be the ATA's DN, so only one person at a time can access voicemail. I spent some time digging into this last year, but came up without an acceptable solution. I may be forgetting or misremembering some of the details but the end result is the same: you can hack something into it but it's a shitty solution. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems to call brazil from germany
Sebastian, This is possible and most likley the reason. To make sure, check the location code of the cause IE in your ISDN disconnect message. You have two options: 1) call your provider and describe your problem. 2) Change your provider Best regards Hans Sebastian Reitenbach schrieb: Hi, my problem in short: I have a problem reaching a co-worker with the asterisk calling from Germany to Brazil. With a chance of about 90% I get a chanunavail message from the asterisk. Normally I try calling him in the afternoon Germany, when he is awake in Brazil. so I tried to make calls to Brazil from Germany via the Asterisk telephone system in the morning, then everything is most likely fine (tried some numbers I found via google) when I do this in the afternoon then I get most likely a all channels unavalable. When I get the all channels unavalable message, then try to call the same number from a mobile, it can reach the number without problem. my theory: I use a cheap preselected carrier to call out via the asterisk, that one has not much overseas lines, and therefore they are getting exhausted at the afternoon, when there are the people up in Brazil. when calling at the same time from a mobile phone, then using another carrier that has more overseas lines and therefore I can reach the number. I have no clue whether my theory is fine, or absolutely stupid. Its just out of the observations I have made so might be completely wrong. so any idea whether my theory is right or not, and if not, any other theories? hope its more clear now. kind regards Sebastian Moises Silva [EMAIL PROTECTED] wrote: Callme stupid, but im not understanding your problem. Suggestions that may help others to answer: 1. A little bit more clear in your examples? :) 2. Try describing the Asterisk behaviour under every circumstance. Regards On 7/17/06, Sebastian Reitenbach [EMAIL PROTECTED] wrote: Hi, I have problems to call to brazil, frome here in germany. the asterisk is connected to the telephone system via a pri interface. I use a preselected provider here to call out. when I try to call a number in brazil, a mobile phone here in the germany in the afternoon, when it is moring in brazil, then the chances to reach that number are next to zero. taking a mobile phone and call that number works fine. when I try to call someone in brazil, taking numbers found by google, then i can reach a lot of these numbers. anybody has an explanation for this? could it be that both carriers have different ways to route the call to brazil and the preselection provider has not so many lines for overseas? kind regards Sebastian -- Sebastian ReitenbachTel.: ++49-(0)3381-8904-451 RapidEye AG Fax: ++49-(0)3381-8904-101 Molkenmarkt 30 e-mail:[EMAIL PROTECTED] D-14776 Brandenburg web:http://www.rapideye.de ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Meridian Tie Line
The BT guy should check LD 73 block LPTI and prompt AFF. If it is crc then you need crc4 as well. Best regards Hans Steve Totaro schrieb: Andy Kirby wrote: I am new to the group but have searched the doc's FAQ's etc before posting here. We are attempting tie our asterisk server/service to the building's PBX, the building is in the UK and the local PBX is a meridian option 11 installed and mainteined by BT. BT Have installed a NTBK50AA E1-PRI card in the meridian with daughter cards NTBK51AA (D channel) and NTAK20BD (Clocking) I have asked BT to configure the card as a Master (Exchange end) E1 Euro ISDN (Just like a standard ISDN30e) They claim to have done this in line with the model they use to interface to Cisco routers etc. I have installed a Digium TE411P in our server looped back the span 2 port (Gives a green light and OK with same config as span 1) and am using a crossover cable to link the PBX to our server. (We tried a pucker BT cross over cable with exactly the same results as mine and a striahgt through gives us nothing at all, I guess as you might expect) I have configured the Zap span for 1 clocking (Primary) 1 line build out, with the framing etc as CCS, HDB3, CRC4 But they don't appear to want to synchronise/talk to each other. ZTTool claims that the span is up and down more times than a fiddlers elbow and the clocking source is internal. ( Might I expect the alarm state to be constant if the framing etc was matched and the clock source to show as external ??) The alarms are cylcling from red to red/yellow and finaly to red/yellow/recover before falling back to red and starting again. I think I may be missing something that is probably blindingly obvious to someone in the know. The BT guy has been very good and is trying to help us get this going but seems rather nonplussed with the terms CRC4, CSS and HDB3... Please can somone help and point me (and I guess by extension the BT guy) in the right direction. Cheers Andy Try your end with every different combination of settings applicable to EuroISDN. I would start by removing the CRC4. Try something, if it doesnt work, put it back and change another setting. s ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FritzCard, mISDN Anlagenanschluss
Yes, it is possible. I'm using PtP and TE mode at home with chan_misdn. Hans Ralf Mueller schrieb: Hello, can someone on the list confirm, that it is possible to connect a FritzCard to an Anlagenschluss, when I use the mISDN driver? I have read a number of posting and articles, that this is not possible with the CAPI driver, but found no clear answer about the mISDN driver. Thanks for your help, Ralf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent
Did you try rtpholdtimeout in sip.conf ? Hans Marco Mouta schrieb: How do I report a Bug to Digium? or asterisk project? On 4/19/06, *Doug Lytle* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Marco Mouta wrote: I've tested maxexpirey=120 and even with this, asterisk didn't stop the call: Scenario: SIP user agent has left without telling to asterisk it was leaving... There was a call to pstn world with MOH running... Any tip to solve this? None. I just confirmed this: Dial from an iax trunk to sip phone SIP phone places call on hold. Unplugged SIP phone Music on hold did not disconnect even after the expiry. Doing a sip show [exten] Shows the peer is unreachable. You may want to file a bug report. Doug ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] reload - restart
Hi ! What is now the difference between a: reload - (cli command reload). restart - (I assume the application asterisk is restarted. o.k starting from new) sip reload - (cli command sip reload). Is sip reload part of the reload command ? Please confirm: Which is the correct command when adding a new SIP party in sip.conf. sip reload ? Thanks ! Hans ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QSIG error -- can somebody explain?
I can only guess, but I think I can remember that the creflen needs to be 2 octets for qsig. Check what the Alcatel switch sends in the setup message to *. Anyway, why do use QSIG ? Does name display work on the * implementation ? Best regards Hans P.S.: Schoene Gruesse an Kurt Krenn Wolfgang Zweimueller schrieb: Hi all, I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the outside world and should forward our calls to the telco. This setup works correctly as far as I use euroisdn as the switchtype. The first problem was that it is only possible to run the * side in CPE-mode -- I wanted NET. Anyway, I configured * this way: switchtype=qsig signalling = bri_cpe facilityenable = yes My experience now is that it is possible to signal a call (both outgoing and incoming) but as soon as the callee takes off the hook the call-setup crashes. Below is the debug log of an outgoing call to a service number of the telco which tells the current time. (The point is that the called number immediately answers the call.) As you can see the Alcatel side answers to our SETUP message with a RELEASE COMPLETE and a cause number 100. This cause (taken from ECMA-143) means: Invalid information element contents , | This cause indicates that the equipment sending this cause has received an | information element which it has implemented; however, one or more of the fields | in the information element are coded in a way that has not been implemented by | the equipment sending this cause. ` Can somebody explain what the problem is? Configuration error, a bug, a problem on the Alcatel-side? Thanks in advance, Wolfgang -- Executing Dial(SIP/1993-567b, Zap/g1/006621503|55|j) in new stack 1 -- Making new call for cr 136 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (8) len=32 1 Call Ref: len= 1 (reference 8/0x8) (Originator) 1 Message type: SETUP (5) 1 [1 041 031 801 901 a31 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [1 181 011 891 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 6c1 061 211 801 311 391 391 331 ] 1 Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number not screened (0) '1993' ] 1 [1 701 0a1 c11 301 301 361 361 321 311 351 301 331 ] 1 Called Number (len=12) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '006621503' ] -- Called g1/006621503 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 8/0x8) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 811 e41 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: (null) (100), class = Protocol Error (6) ] 1 -- Making new call for cr 32776 1 -- Processing IE 8 (cs0, Cause) 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null 1 No response to SETUP message 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending 1 Protocol Discriminator: Q.931 (8) len=8 1 Call Ref: len= 1 (reference 8/0x8) (Originator) 1 Message type: DISCONNECT (69) 1 [1 081 021 811 921 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Unknown (18), class = Normal Event (1) ] -- Channel 0/1, span 1 got hangup, cause 42 -- Zap/1-1 is circuit-busy 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Request, peerstate Disconnect Indication -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Busy(SIP/1993-567b, ) in new stack == Spawn extension (dialout, 436621503, 102) exited non-zero on 'SIP/1993-567b' 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 8/0x8) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 811 d11 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Unknown (81), class = Invalid message (5) ] 1 -- Making new call for cr 32776 1 -- Processing IE 8 (cs0, Cause) 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null,
Re: [Asterisk-Users] ATA's and faxing
Enable pass thru fax mode on the HT486, or enable ulaw in your SIP config. Hans Garth van Sittert schrieb: Hi All Is there any special configuration needed to send and receive faxes on an ATA device? I am using G711.a with a Grandstream Handytone 486. I can send faxes from a fax machine on the ATA, but receiving doesn't work. I get the fax signal, but it just doesn't continue. The LAN is used purely for VoIP traffic. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's and faxing
ulaw was neccessary when pass through was disabled. What does a sip debug tell you ? Hans Garth van Sittert schrieb: I am using alaw and I have already enabled the pass through. Does alaw and ulaw work? I can fax out, but not receive faxes. Garth Johann Steinwendtner wrote: Enable pass thru fax mode on the HT486, or enable ulaw in your SIP config. Hans Garth van Sittert schrieb: Hi All Is there any special configuration needed to send and receive faxes on an ATA device? I am using G711.a with a Grandstream Handytone 486. I can send faxes from a fax machine on the ATA, but receiving doesn't work. I get the fax signal, but it just doesn't continue. The LAN is used purely for VoIP traffic. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No translator path: iax2 calls not possible
Hello ! Asterisk 1.0.9 running on Linux 2.6.12. I'm not able to call iax2 channels. There can be no translation path found. When I try to call from a ZAP PRI channel the following error occurs: channel.c:1891 ast_request: No translator path exists for channel type IAX2 (native 63488) to 72 dial_exec: Unable to create channel of type 'IAX2' What is wrong ? Here is my iax.conf: [general] bindport=4569 language=de disallow=all jitterbuffer=no tos=lowdelay [iax-test1] type=friend username=iax-test1 secret=xx context=from_IAX host=dynamic qualify=yes allow=alaw allow=ulaw Thanks for your help ! Hans ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_misdn crashes : init_stack: success but entitylist not empty
Make sure that you compile misdnuser with gcc3.x, gcc4 did not work for me. Hans Yoann Le Bihan schrieb: Jose, I met so many problems these last 8 days that I don't remember exactly which config was mine at that time, so I can't testify the answer... (just for fun : my linux box is having 3 hd with a different distro on each of them and I plug the cable on the hd I want to boot depending on my mood ;o)). I think I was running 1.0.9. The main things I did were : - deinstalling everything (asterisk, misdn, misdnuser, chan_misdn, ...) - compiling and installing asterisk 1.2.0 (make ; make install) - downloading the install_misdn script on beronet (http://www.beronet.com/download/install-misdn.tar.gz) and executing the make install (be careful : you need kernel headers) And now, I'm done : Asterisk runs without chan_misdn, but crashes with it :-( But it's installed :-) Good luck ! ;) Cheers, YLB. 2005/11/25, Jose Limeres [EMAIL PROTECTED]: Yoann, I am going through a similar problem you reported in a past posting: Nov 24 17:49:31 ERROR[9326] chan_misdn.c: Unable to initialize mISDN Nov 24 17:49:31 WARNING[9326] loader.c: chan_misdn.so: load_module failed, returning -1 Nov 24 17:49:31 WARNING[9326] chan_misdn.c: cb_log called with out-of-range port number! (0) Nov 24 17:49:31 WARNING[9326] loader.c: Loading module chan_misdn.so failed! How did you solve it? Thanks, jose On 25/11/05, Yoann Le Bihan [EMAIL PROTECTED] wrote: Hi, Asterisk 1.2 on FC4, all is right, I'm happy. But when I try to load chan_misdn after a successful install, I get it : # asterisk -vvvgc [...] [chan_features.so] = (Feature Proxy Channel) == Registered channel type 'Feature' (Feature Proxy Channel Driver) [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found Got: 1 from get_ports Init. Stack on port:1 No Connect port:1 init_stack: Success # Nothing else. Asterisk crashes. If I look at /var/log/messages : # tail /var/log/messages Nov 25 00:22:39 toto kernel: Debug: sleeping function called from invalid context at arch/i386/lib/usercopy.c:634 Nov 25 00:22:39 toto kernel: in_atomic():0, irqs_disabled():1 Nov 25 00:22:39 toto kernel: [c01d16b8] copy_from_user+0x18/0x80 Nov 25 00:22:39 toto kernel: [e0ba85b8] mISDN_write+0x318/0x7c5 [mISDN_core] Nov 25 00:22:39 toto kernel: [e0ba82a0] mISDN_write+0x0/0x7c5 [mISDN_core] Nov 25 00:22:39 toto kernel: [c0158bb1] vfs_write+0xa2/0x15a Nov 25 00:22:39 toto kernel: [c0158d14] sys_write+0x41/0x6a Nov 25 00:22:39 toto kernel: [c0102ec5] syscall_call+0x7/0xb Nov 25 00:22:39 toto kernel: MISDN free_device: entitylist not empty # Any idea ?... I've been on it for 1 whole week... I'm exhausted :-( Cheers, YLB. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mission-Critical Deployments
John, why don't you migrate slowly to asterisk ? If you want to keep most of your analog phone hardware, leave it on your Meridian 1. The M1 is doing a good job on features on analog phone sets. Also, your users are familiar with the call handling of the M1. Install VoIP phones on Asterisk and connect Asterisk via PRI to the M1. Then you can replace step by step your phones. Regards, Hans John Goerzen schrieb: I work for a company that is nearing the end-of-life on its existing Nortel Meridian switch and is considering Asterisk. We have approximately 200 existing extensions, and probably 150 out of those 200 are using basic analog phones and would stay that way. The rest would have VOIP phones at the desk. We're seriously considering switching to Asterisk. I've done quite a bit of tinkering with Asterisk for my home, but I'm not certain about a few aspects of how we might deploy Asterisk in the enterprise. Here are my questions: 1. Where could I look for some resources on server sizing? Is it any problem to support this number of users with a single server? 2. What do we need to do for our data network to make VOIP reliable? QoS, basic traffic prioritization on the switch, vlan, ??? 3. What's the best way to integrate these 150 analog extensions? I've seen interface boxes that usually come in 24-port sizes. Some have an Ethernet/SIP interface to hook up to Asterisk, and others have a T1 interface. What sounds best and is the most reliable? 4. What is a good company to contract with for emergency support? Digium? 5. What are people doing to make VOIP phones resiliant in the face of power outages? Is there anybody here that would be willing to serve as a reference check for Asterisk should we pursue that path? Thanks, -- John ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI E1 Problem only chan 17-31
Another problem could be that there is a B-channel mismatch. e.g. Asterisk uses channel 26 and Nortel uses channel 25. This can be modified on at least QSIG trunks. But on EuroISDN there should not be a problem. Hans [EMAIL PROTECTED] schrieb: On Wed, 2 Nov 2005, Alvaro Parres wrote: Hi list I have a problem on a PRI E1 card. The connection diagram is: [ASTERISK] -- PRI-NET -- PRI-CPE -- [NORTELL] The problem is: When i made a call in channels 17 to 31, there is no voice in any way... but on channels 1 to 15 i have no problems... The Nortel end only has 15 channels enabled?; probably they want you to pay more to enable more? Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 with Asterisk
Goran, which company ist this ? Do they use the www.ss7box.com approach ? Thanks and best regards Hans Goran Skular schrieb: anyone running SS7 with Asterisk ? Please help me out. I need to know the hardware used for SS7 with Digium E1 cards... I can point you to one company in Austria. They deployed SS7 on Asterisk, but not with Digium cards for one smaller telco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc problem: overlapdial don't work after update bristuff
I had the same problem. It seems that a fix into bristuff for .at
does not work very well.
I 've patched chan_zap.c
Best regards
Hans
old:
} else {
if (pri-nodetype == BRI_CPE) {
/* fix for .at p2p bri lines */
pri-pvts[chanpos]-exten[0] = 's';
new:
} else {
if ((pri-nodetype == BRI_CPE)
(!pri-overlapdial)) {
/* fix for .at p2p bri lines */
pri-pvts[chanpos]-exten[0] = 's';
[EMAIL PROTECTED] schrieb:
hello,
I have a asterisk box (Slackware 9.1.0, Linux 2.4.31) connected to a
Ericsson Businessphone PBX on the internal S0 bus with HFC-S card and
zaphfc driver - point2point mode -.
-
| TELCO |
| BRI |
-
|
| PBX external S0
| PBX |
| PBX internal P2P S0 NT Mode
|
| HFC-S Card P2P TE Mode
-
| * |
| BOX |
-
Since nearly one year it runs Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b
fine. after dialing the internal S0 number (61) I get overlapdial and
dialtone.
[example dial the PBX S0 with Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b]
-- Accepting AUTHENTICATED call from 192.168.255.1, requested format
= 8, actual format = 8
-- Executing Dial(IAX2/[EMAIL PROTECTED]/1, Zap/g1/61|60) in new stack
-- Called g1/61
Sep 23 10:27:50 NOTICE[229390]: chan_zap.c:8725 pri_dchannel: SETUP_ACK
for 'Zap/1-1'
-- Accepting overlap call from '' to 'unspecified' on channel 0/2,
span 1
-- Starting simple switch on 'Zap/2-1'
here I get a dialtone from PBX and I can dial
start dialing for 8 seconds
for different reasons I tried to update on latest bristuff-0.2.0-RC8n.
after update the zaphfc overlap dial don't work anymore.
when I dial the S0 bus it jumps directly into 's' extension,
which does not exisist in the context and I get 'no number' from PBX.
zapata.conf:
[channels]
switchtype = euroisdn
signalling = bri_cpe
pridialplan=local
echocancel=yes
immediate=no
overlapdial=yes
group = 1
faxdetect=both
context=fromisdn
channel = 1-2
zaptel.conf:
loadzone=nl
defaultzone=nl
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
extension.conf:
[fromisdn]
; example dial to BRI on other location
exten = _0.,1,Dial(IAX2/call_saar/[EMAIL PROTECTED],60)
exten = _0.,2,Congestion
exten = _0.,102,Hangup
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Re: [Asterisk-Users] pri release cause code mismatch
Tirpák Miklós schrieb: Yes. 34 is required by the Nortel to send the call to an alternative destination. Cause 38 or 42 triggers the rerouting also for both options. Hans ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pri release cause code mismatch
Hi ! Asterisk sends a RELASE COMPLETE with cause code 34. It seems that Nortel expects a RELEASE message in this state. The conversion is done in the protocol engine of the MSDL. Why would you want the cause code 34 to be sent ? Do you need a special rerouting on the Nortel side ? Would it be a help if you send a cause 3 ? (RELASE msg) Best regards Hans Tirpák Miklós schrieb: Hi! My asterisk (1.0.7) is connected to a Nortel pbx with Digium E100P card, both side are ETSI EuroISDN. I would like to reject an incomming call with cause code 34, but the Nortel PBX gets the value of 31 instead of 34. It seems to work on the asterisk side: Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 17162/0x430A) (Originator) Message type: SETUP (5) ... Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 49930/0xC30A) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 a2] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Circuit/channel congestion (34), class = Network Congestion (2) ] My macro looks like: exten = s,1,SetVar(PRI_CAUSE=34) exten = s,2,Hangup According to the debug on Nortel it gets 31 cause code in the release complete q.931 message. Do you have any idea? Thanks, Miklos ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk + chan_mISDN = undefined symbol: ast_pickup_call
Christian Wengel schrieb: Hi! I tried install-misdn.tgz from http://www.beronet.com/download/ , some minutes ago. Also I switched to an older kernel (2.6.8), but I get the same error. I think that I made the correct changes in the Makefiles, but I will attach them to this e-mail, maybe you see something wrong. Is there a change when you uncomment this flag ? # ASTERISK Version # If you are using a asterisk version above from stable (v1-0) # then comment the following line out (good luck) # #CFLAGS+=-DASTERISK_STABLE Best regards Hans ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk + chan_mISDN = undefined symbol: ast_pickup_call
This error would occur if you compile chan_misdn against Asterisk stable and not specifying it in the makefile of chan_misdn. Check the makefile of chan_misdn. Hans Christian Wengel schrieb: Hi all! I'm getting an error when I try to start asterisk with chan_misdn. I patched my kernel (2.6.12.4), and compiled the whole stuff (kernel, mISDNuser, asterisk, chan_misdn). I got mISDN from http://isdn.jolly.de/download/v3.0/ I'm using a CVS Snapshot of asterisk, which was checked out about 5 hours ago. This is the error: [chan_misdn.so]Aug 15 14:13:29 WARNING[4929]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/chan_misdn.so: undefined symbol: ast_pickup_call Aug 15 14:13:29 WARNING[4929]: loader.c:488 load_modules: Loading module chan_misdn.so failed! I have no idea where to start solving this problem. Has anybody a hint for me? If you need more information, feel free to ask for it. Greets, Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MISDN callerid
This chan_misdn version is old, use a newer one. It seems that TypeOfNumber interpretation has not been integrated in this verison. Best regards Hans Christian Peter schrieb: Hi all, I have a cologne chip card which is connected directly to the ntba. Outgoing and incoming calls work fine, but incoming calls from ntba have the wrong callerid (first 0 is missing). I'm using current jolly misdn drivers and chan_misdn-14_04_05 with asterisk stable. Is anyone seeing this behaviour too? Thanks in advance Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Voice mail to nortel PBX Option 11
Isn't it possible to turn on MWI via background terminal ? In that case an application needs to do this via serial interface. best regards Hans Users will have to get into the habit of calling the VM to check if there's messages. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN DID
There is no called party ie but sending complete ie included in the setup message. Hence, it tries to terminate. Best regards Hans Paul Belanger schrieb: Where are your calls coming from? Are you connected to the Telco or PBX? PB Panitaxx wrote: Hi, thanks for your response. here is the log of one call: Enabled debugging on span 1 Asterisk*CLI Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: SETUP (5) [a1] Sending Complete (len= 1) [04 03 90 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 8d] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 13 ] [1e 02 84 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0b 00 83 39 31 35 34 35 31 39 30 30] Calling Number (len=13) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation allowed of network provided number (3) '915451900' ] -- Making new call for cr 72 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, C alling Party Number) -- Going to extension s|1 because of Complete received Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8d] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 13 ] -- Accepting call from '915451900' to 's' on channel 0/13, span 1 Asterisk*CLI -- Executing Playback(Zap/13-1, vm-intro|noanswer) in new stack Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: PROGRESS (3) [1e 02 81 88] 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: Inband information or appropriate pattern now available. (8) ] -- Playing 'vm-intro' (language 'es') Asterisk*CLI -- Executing Playback(Zap/13-1, vm-goodbye) in new stack Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: CONNECT (7) [18 03 a9 83 8d] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 13 ] [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) ] -- Playing 'vm-goodbye' (language 'es') Asterisk*CLI Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Executing NoOp(Zap/13-1, ) in new stack -- Executing Hangup(Zap/13-1, ) in new stack == Spawn extension (primario, s, 4) exited non-zero on 'Zap/13-1' NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/13-1' Asterisk*CLI On 8/9/05, jj [EMAIL PROTECTED] wrote: What does pri debug span 1 show? On Aug 9, 2005, at 5:02 PM, Panitaxx wrote: Hello, I have an ISDN PRI E1. For some reason I am not receiving the did number so every call can only go to s exten. I have tried using _X. exten. Also I have immediate=no in zapata.conf. Any hint? thanks in advance, Iván Aponte ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
[Asterisk-Users] Cisco IP Phones on Asterisk: chan_sip or chan_sccp
Hello ! I 'd like to connect Cisco IP phones to *. (7940 7960) Shall I use SIP or SCCP. Which approach provides better support of features of the Cisco IP phones ? Thanks ! Johann ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No rering on misoperation on SIP ATA
Hello ! Following scenario: Party A: SIP Analog Terminal Adapter Grandstream HT486 (analog phone) Party B: any other external PSTN set Asterisk 1.0.9 Party A calls external party. Call is established. Party A presses the flash key and goes on hook. The external Party still gets Music on Hold. No disconnection. I would have expected that Party A would rering. Is this a problem of the Grandstream Adapter or is this a problem of Asterisk ? Hans ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic D/300/SC-2E1
Hello ! I got a dual E1 card from Dialogic (D300/SC-2E1 old card with ISA) at my desk. Is there a channel driver available for this kind of card ? Best regards Johann ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fritz PCI card in ptp mode with chan_misdn
Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. Afer compiling mISDN into the kernel and building chan_misdn Asterisk stops loading with : [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables the asterisk to use hardware which is supported by the new ) debug_init: using stdout for debug log debug_init: using stderr for warning log debug_init: using stderr for error log debug_init: debug_mask = 0 Locking Config Mutex UnLocking Config Mutex Init. Stack on port:1 unknown port(1) type 0x No lower Id port:1 init_stack: Success aster1:/etc/asterisk # But the above message init_stack: Success is not true. The mISDN stuff is loaded as follows: modprobe mISDN_core modprobe mISDN_l1 debug=0 modprobe mISDN_l2 debug=0 modprobe l3udss1 modprobe mISDN_dsp debug=0 options=0x0 modprobe avmfritz layermask=0xf protocol=0x22 The relevant misdn.conf show: ; chan_misdn sample config ; general section: ; [general] ; define your default context here context=misdn ; default language language=en ; set debugging flag: ; 0 - No Debug ; 1 - mISDN Messages and * - Messages, and * - State changes ; 2 - Messages + Message specific Informations (e.g. bearer capability) ; 3 - very Verbose, the above + lots of Driver specific infos ; 4 - even more Verbose than 3 ; debug=3 ; ; Stops dialtone after getting first digit on nt Port ; stop_tone_after_first_digit=yes ; ; Wether to append overlapdialed Digits to Extension or not ; append_digits2exten=yes ; users sections: ; ; name your sections as you which but not general ! ; the secions are Groups, you can dial out in extensions.conf ; with Dial(mISDN/g:extern/101) where extern is a section name, ; chan_misdn tries every port in this section to find a ; new free channel ; [extern] ; define your ports, e.g. 1,2 (depends on mISDN-driver loading order) ports=1 ; context where to go to when incoming Call on one of the above ports context=from_meridian ; uncomment the following to get into s extension at extension conf ; there you can use DigitTimeout if you can't or don't want to use ; isdn overlap dial. ;immediate=yes ; uncomment the following to have hold and retrieve support ;hold_allowed=yes ; Group Specific Language language=en ; indicate the incoming calls to asterisk ; here you can give a comma seperated list or simply an '*' for ; any msn. ;msns=* ; here an example with given msns ;[second_extern] ;ports=5 ;context=Extern2 ;msns=102,144,101,104 Linux Debian Kernel 2.6.12 Thanks for any hints ! Best regards Hans ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
