Re: [asterisk-users] chan_capi audio weirdness
Hi Armin, On 18/02/2012 19:28, Arik Raffael Funke wrote: in NT mode, the B-channel is not activated automatically. You have to signal the TE side that early-B3 data is available. Then the TE side can activate the B-channel. If the NT-side is chan_capi, use exten = _X.,n,capicommand(progress) I tried what you suggested - but without any luck. To make sure I did not misunderstand you, I now have: exten = _X.,1,capicommand(progress) exten = _X.,n,Dial(CAPI/capi_intern/12345/b) To help identification of the problem, my console prints the following after capicommand(progress): [Feb 19 10:45:08] WARNING[3483]: chan_capi.c:4972 show_capi_conf_error: ISDN_INTERN#02: conf_error 0x2001 PLCI=0x103 Command=SELECT_B_PROTOCOL_CONF,0x8495 ISDN_INTERN#02: CAPI INFO 0x2001: Message not supported in current state Regards, Arik -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] chan_capi audio weirdness
Hello Armin, On 15/02/2012 22:53, Armin Schindler wrote: I hear no progress indication. EVEN when using the r-option of the dial command. It works however with exten = _X.,1,Answer exten = _X.,n,Dial(CAPI/contr1/12345) in NT mode, the B-channel is not activated automatically. You have to signal the TE side that early-B3 data is available. Then the TE side can activate the B-channel. If the NT-side is chan_capi, use exten = _X.,n,capicommand(progress) without Answer before Dial(). Also, when using Dial() with chan_capi, you should use /b or /B option in Dial() to get early-B3 from that other side too. See README of chan_capi package for more details. Thank you for your help. I did look at the chan_capi README, but I am afraid I do not know enough about the capi protocol to make sense of everything. I tried what you suggested - but without any luck. To make sure I did not misunderstand you, I now have: exten = _X.,1,capicommand(progress) exten = _X.,n,Dial(CAPI/capi_intern/12345/b) Used to call either from intern-intern or alternatively intern-extern. In neither case do I get progress indication (i.e. ringing tones) as I did when answering the channel before dialing. Could it be that my hardware is simply behaving unexpectedly? After all, it's not really a traditional capi card but an embedded device. Or am I still doing something wrong? Many thanks, Arik -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] chan_capi audio weirdness
Hi, I am trying to run asterisk on an AVM Fritz!Box Fon 7270 embedded DSL router. This works quite well after getting rid of the preinstalled phone server but I am encountering some unexpected behaviour. Background: I am using two CAPI controllers provided by the hardware - one in MSN mode for dialling out and - one in NT-mode, (DID) for the internal S0-Bus The problem is, I get no audio whatsoever until a channel is answered. Some of the symptoms of this are: - If I have an s-extension for the internal S0-Bus exten = s,1,Playtones(dial) I cannot hear the dialtone. It works however with: exten = s,1,Answer exten = s,n,Playtones(dial) - Similarly if I dial from internal to external with the extension: exten = _X.,1,Dial(CAPI/contr1/12345) I hear no progress indication. EVEN when using the r-option of the dial command. It works however with exten = _X.,1,Answer exten = _X.,n,Dial(CAPI/contr1/12345) Has anybody seen this before? Many thanks, Arik -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Call holding with chan_capi
Hi, I am using ISDN phones which have a Park call button. The idea is: you are on a call, push the button and hang up. You can then go to another phone and pickup the call without having to remember parking slots, etc. Unfortunately I cannot figure out how to get it to work with asterisk. I suspect it has something to do with capicommand(holdtype|local)... Does anybody use this isdn functionality with asterisk? Many thanks, Arik -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call holding with chan_capi
In case this helps, when pressing the Park Call button, I get the following with capi debug: DISCONNECT_REQ ID=002 #0x037e LEN=0013 Controller/PLCI/NCCI= 0x1303 AdditionalInfo = default CAPI: ApplId=0x0002 Command=0x04 SubCommand=0x81 MsgNum=0x037e NCCI=0x1303 DISCONNECT_CONFID=002 #0x037e LEN=0014 Controller/PLCI/NCCI= 0x1303 Info= 0x0 CAPI: ApplId=0x0002 Command=0x04 SubCommand=0x82 MsgNum=0xe3a5 NCCI=0x1303 DISCONNECT_IND ID=002 #0xe3a5 LEN=0014 Controller/PLCI/NCCI= 0x1303 Reason = 0x0 DISCONNECT_RESPID=002 #0xe3a5 LEN=0012 Controller/PLCI/NCCI= 0x1303 On 14/02/2012 18:18, Arik Raffael Funke wrote: Hi, I am using ISDN phones which have a Park call button. The idea is: you are on a call, push the button and hang up. You can then go to another phone and pickup the call without having to remember parking slots, etc. Unfortunately I cannot figure out how to get it to work with asterisk. I suspect it has something to do with capicommand(holdtype|local)... Does anybody use this isdn functionality with asterisk? Many thanks, Arik -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call holding with chan_capi
My apologies, I just realised I copied the wrong section of the debug log. So once again, when pressing the park call button, I get the following capi debug output: CAPI: ApplId=0x0002 Command=0x80 SubCommand=0x82 MsgNum=0xe446 NCCI=0x1403 FACILITY_IND ID=002 #0xe446 LEN=0018 Controller/PLCI/NCCI= 0x1403 FacilitySelector= 0x3 FacilityIndicationParameter = 02 80 00 -- ISDN_INTERN#02: unhandled FACILITY_IND supplementary function 8002 FACILITY_RESP ID=002 #0xe446 LEN=0015 Controller/PLCI/NCCI= 0x1403 FacilitySelector= 0x3 FacilityResponseParameters = default CAPI: ApplId=0x0002 Command=0x84 SubCommand=0x82 MsgNum=0xe447 NCCI=0x00011403 DISCONNECT_B3_IND ID=002 #0xe447 LEN=0015 Controller/PLCI/NCCI= 0x11403 Reason_B3 = 0x3301 NCPI= default DISCONNECT_B3_RESP ID=002 #0xe447 LEN=0012 Controller/PLCI/NCCI= 0x11403 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] mISDN Asterisk 1.4: HFC-S card not responsive
Hi, I have installed Asterisk 1.4 with mISDN with the install-asterisk.tar.gz script from beronet.com. On my system I have two cards, one a AVM Frit!Card Pci 2.0 and one HFC-S chip. I know both to work well with mISDN on my system from a previous installation. Now however, the AVM card works well at first glance, i.e. it registers incoming calls and works through the asterisk dialplan. Calls on the hfc card however seem to be completely ignored. There is not the slightest indication in asterisk that call come in. The CLI stays completely silent even for debug and verbose levels of 100 for core and misdn! The HFC-S card however does seem to be not completely ignored by asterisk: if I plug-in or remove connection with a high misdn debug level, it shows the usual messages - as it also does for the AVM card. Only incoming calls are ignored - n.b. outgoing do not work either... Below are outputs from the CLI (misdn show config, misdn show stacks, pluggin in cable, removing cable), dmesg and lspci -v. I hope somebody could give me a hint as to what could be the problem. The system is freshly installed and both cards are configured identically. Cheers, Arik = CLI: misdn show config = (n.b. port 1=hfcpci; port 2=avmfritz) *CLI misdn show config Misdn General-Config: - misdn_init: /etc/misdn-init.conf - debug: 0 - tracefile: /var/log/asterisk/misdn.log - bridging: no - stop_tone_after_first_digit: yes - append_digits2exten: yes - dynamic_crypt: no- crypt_prefix: ** - crypt_keys: test,muh - ntdebugflags: 0 - ntdebugfile: /var/log/misdn-nt.log [PORT 1] - name: intern - allowed_bearers: all - far_alerting: no - rxgain: 0 - txgain: 0- te_choose_channel: no - pmp_l1_check: no - reject_cause: 16 - block_on_alarm: no - hdlc: no - context: Intern - language: en - musicclass: default - callerid: - method: standard - dialplan: 0 - localdialplan: 0 - cpndialplan: 0 - nationalprefix: 0- internationalprefix: 00 - presentation: -1 - screen: -1 - always_immediate: no - nodialtone: no - immediate: no- senddtmf: yes - hold_allowed: no - early_bconnect: yes - incoming_early_audio: no - echocancel: 0 - need_more_infos: no - noautorespond_on_setup: no - nttimeout: no- bridging: yes - jitterbuffer: 4000 - jitterbuffer_upper_threshold: 0 - callgroup: - pickupgroup: - max_incoming: -1 - max_outgoing: -1 - l1watcher_timeout: 0 - overlapdial: 0 - msns: * - faxdetect: no - faxdetect_context: - faxdetect_timeout: 5 - ptp: no [PORT 2] - name: intern - allowed_bearers: all - far_alerting: no - rxgain: 0 - txgain: 0- te_choose_channel: no - pmp_l1_check: no - reject_cause: 16 - block_on_alarm: no - hdlc: no - context: Intern - language: en - musicclass: default - callerid: - method: standard - dialplan: 0 - localdialplan: 0 - cpndialplan: 0 - nationalprefix: 0- internationalprefix: 00 - presentation: -1 - screen: -1 - always_immediate: no - nodialtone: no - immediate: no- senddtmf: yes - hold_allowed: no - early_bconnect: yes - incoming_early_audio: no - echocancel: 0 - need_more_infos: no - noautorespond_on_setup: no - nttimeout: no- bridging: yes - jitterbuffer: 4000 - jitterbuffer_upper_threshold: 0 - callgroup: - pickupgroup: - max_incoming: -1 - max_outgoing: -1 - l1watcher_timeout: 0 - overlapdial: 0 - msns: * - faxdetect: no - faxdetect_context: - faxdetect_timeout: 5 - ptp: no *CLI = CLI: misdn show stacks = *CLI misdn show stacks BEGIN STACK_LIST: * Port 1 Type TE Prot. PMP L2Link DOWN L1Link:UP Blocked:0 Debug:1 * Port 2 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:1 *CLI = CLI: when plugging in hfc card = *CLI misdn set debug 100 changing debug level for all ports to 100 *CLI *CLI P[ 0] Got empty Msg.. P[ 0] MGMT: Short status dinfo 101 P[ 0] MGMT: SSTATUS: L1_ACTIVATED P[ 0] Got empty Msg.. *CLI = CLI: removing cable from hfc card = *CLI P[ 0] Got empty Msg.. P[ 0] MGMT: Short status dinfo 100 P[ 0] MGMT: SSTATUS: L1_DEACTIVATED P[ 1] $$$ find_chan: No channel found for oad: dad: P[ 0] get_index: event not found! P[ 1] I IND :CLEAN_UP oad: dad: pid:0 state:none P[ 1]
[asterisk-users] DTMF auto detection bug?
Hi, it seems that there is a bug in asterisk's dtmf mode autodetection. Assume following sip.conf: [sipprovider] disallow=all allow=g726 dtmfmode=auto DTMF does not work. It seems rfc2833 mode is chosen despite it being obvious that this cannot work! The following configuration is necessary to get DTMF to work: dtmfmode=info In my opinion, this behaviour is counter-intuitive. I am using asterisk 1.2. In v. 1.4 does dtmfmode=auto still have the behaviour? Cheers, Arik ___ --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] Re: DTMF auto detection bug?
Joshua Colp wrote: Arik Raffael Funke wrote: The auto setting also does not encompass the info DTMF option for sending. Thanks. I was not aware of this. Ragards, - Arik ___ --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] Ringing does not terminate on mISDN after pickup
Hello, I am having something of an odd problem: about every 100 calls or so, when a call comes in via an external mISDN interface and I route it to an internal mISDN interface by dialing an internal msn that is programmed for multiple phones on the internal bus, somtimes the other phones continue ringing for several minutes after the call has already been picked up by one (or even eventually hungup already...). As you imagine this is really annoying... I don't seem to be able to narrow it down in any way: show channels is empty (provided, call was answered an hungup, other phones continue ringing) The only way to terminate the ringing is: wait for several minutes... :-) or: restart asterisk. Can anybody point me in the right direction with this issue? Every attempt to get rid of the problem has failed. I have no clue what else I could try... - Does anybody else experience this as well? Cheers, Arik ___ --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] Occasional SMS problem
Hi, I am using asterisk's SMS functionality for sending messages. Most of the time it works without problems (as in situation 1) but sometimes something seems to go wrong with the transmission (as in situation 2). I am using Deutsche Telekom, Germany's main TELCO, so I suppose the problem must be on my end. Can anybody tell me what is going on or how I could narrow down the problem? Cheers, Arik Situation 1 === -- Attempting call on mISDN/g:extern/0193010 for application SMS(0) (Retry 1) funke*CLI funke*CLI Channel mISDN/2-u11 was answered. Launching SMS(0) on mISDN/2-u11 -- SMS RX 93 00 6D -- SMS TX 91 1C 01 0B 0C 81 70 21 41 43 58 03 00 F1 11 C4 74 79 0E 4A CF E9 A0 72 DA 0D A2 96 E7 74... -- SMS RX 95 09 01 00 70 20 82 12 55 70 40 38 -- SMS TX 94 00 6C Feb 28 21:55:09 NOTICE[1963]: pbx_spool.c:279 attempt_thread: Call completed to mISDN/g:extern/0193010 Situation 2 === -- Attempting call on mISDN/g:extern/0193010 for application SMS(0) (Retry 1) Channel mISDN/2-u10 was answered. Launching SMS(0) on mISDN/2-u10 -- SMS RX 93 00 6D -- SMS TX 91 1C 01 0A 0C 81 70 21 41 43 58 03 00 F1 11 C4 74 79 0E 4A CF E9 A0 72 DA 0D A2 96 E7 74... -- SMS RX 95 09 01 00 70 20 82 12 25 84 40 54 -- SMS TX 94 00 6C -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E Feb 28 21:53:28 NOTICE[1872]: pbx_spool.c:279 attempt_thread: Call completed to mISDN/g:extern/0193010 ___ --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] Has anybody voipstunt working?
Hi, I wrote a few days ago about my problem with calls via voipstunt stopping ringing after 5-6 rings, subj. SIP Dial out timeout. Though if the remote station picks up before that, everything works flawlessly. I am not entirely sure when this phenomenon popped up, but I used the this configuration for quite some time before without any hassles. Nobody else seems to have this problem. So I guess it nevertheless has something to do with my configuration... Could anybody who has voipstunt working to their satisfaction send me their configuration files via PM? [EMAIL PROTECTED] Thanks a lot! - Arik PS: Just a thought: Has something about the sip.conf changed between v1.0 and 1.2 that might be related? I am essentially using my sip.conf from v1.0. Or in any other configuration file? ___ --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] Re: SIP Dial out timeout
Eric ManxPower Wieling wrote: Arik Raffael Funke wrote: I am having a problem that is a miracle to me: If I dial out via voipstunt.com the call rings for a few seconds and then gives me a busy sign. Start out with not using the r option to the Dial line. That will remove the faked ringing tone. I am not having one in my dial string: -- Executing Dial(mISDN/1-1, SIP/[EMAIL PROTECTED]||Tt) in new stack Any other ideas? BTW: I am not sure I made this clear: The remote station does actually ring. I am not just getting a ringing indication. - Only the ringing stops after about 6 rings Regards, Arik ___ --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] SIP Dial out timeout
Hi, I am having a problem that is a miracle to me: If I dial out via voipstunt.com the call rings for a few seconds and then gives me a busy sign. - I do not have a timeout set in my dial command - the remote station does not cause the busy either - dialing the number with the voipstunt client does not give busy after a few seconds - dialing with the same voipstunt account with a softphone works without problems - when dialing out via other channels, i.e. iax or misdn on the asterisk machine, no timeout problem -in the CLI there is no message at all when the timeout occurs. It shows: ...snipp... -- Called [EMAIL PROTECTED] -- SIP/voipstunt-081b61a8 is making progress passing it to mISDN/1-1 P[ 1] After SETUP BC funke*CLI Given these facts I believe that the problem has something to do with my asterisk setup, and more specifically, as it only occurs with SIP, sip.conf. Is that reasonable? Unfortunately I have absolutely no idea, how to narrow it down further. My sip.conf looks as follows: -- [general] bindport=5060 bindaddr=0.0.0.0 srvlookup=yes allowguest=no qualify=no srvlookup=yes canreinvite=yes [voipstunt] type=friend host=sip.voipstunt.com disallow=all allow=g726 username=my_account fromuser=my_account secret=my_password qualify=2000 canreinvite=no promiscredir=yes rtptimeout=300 rtpholdtimeout=300 -- If anybody has any hint on how this might be solved, please let me know. Cheers. Arik ___ --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] Re: mIDN question
Hi,
I found a solution to my own problem... well, sort of.
When I put the follwing in my misdn.conf:
[intern]
ports=1
callgroup=1
pickupgroup=1
always_immediate=yes
nodialtone=yes
context=intern
I.e. send misdn_chan directly to the s extension and do not allow it to
create a dial tone, and instead create the dialtone in extensions.conf
as follows:
[intern]
exten = s,1,Playtones(dial)
exten = _X.,1,Goto(dial,${EXTEN},1)
everything works as it should.
Though I don't understand why I need to create the dial tone manually
just to get the digit timeout to be taken into account by asterisk...
Kind regards,
Arik
Arik Raffael Funke wrote:
Hi,
I have switched a while back from chan_capi to chan_misdn. When the
number is dialed and the phone is then picked up everything works just
fine. Some users however FIRST pick up the phone and then start to
dial... I did not get this to work with misdn.
When two digits have been dialed, asterisk sees the extension as
complete and does not wait for further digits. I am using an midsn NT
port that feeds into following dialplan context:
[intern]
exten = _X.,1,Macro(dial)
How is this done properly with misdn?
Thanks,
Arik
PS: I am using the following options in misdn.conf (basically they are
the defaults):
[general]
misdn_init=/etc/misdn-init.conf
debug=0
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
bridging=no
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh
[default]
context=misdn
language=de
musicclass=default
senddtmf=yes
far_alerting=no
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=no
pmp_l1_check=yes
need_more_infos=no
method=standard
dialplan=0
localdialplan=4
cpndialplan=0
early_bconnect=no
incoming_early_audio=no
nodialtone=no
presentation=-1
screen=-1
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
hold_allowed=yes
[intern]
ports=1
callgroup=1
pickupgroup=1
context=intern
[extern]
ports=2
context=extern
msns=*
echocancel=128
___
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[asterisk-users] mIDN question
Hi, I have switched a while back from chan_capi to chan_misdn. When the number is dialed and the phone is then picked up everything works just fine. Some users however FIRST pick up the phone and then start to dial... I did not get this to work with misdn. When two digits have been dialed, asterisk sees the extension as complete and does not wait for further digits. I am using an midsn NT port that feeds into following dialplan context: [intern] exten = _X.,1,Macro(dial) How is this done properly with misdn? Thanks, Arik PS: I am using the following options in misdn.conf (basically they are the defaults): [general] misdn_init=/etc/misdn-init.conf debug=0 ntdebugflags=0 ntdebugfile=/var/log/misdn-nt.log bridging=no stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh [default] context=misdn language=de musicclass=default senddtmf=yes far_alerting=no allowed_bearers=all nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 te_choose_channel=no pmp_l1_check=yes need_more_infos=no method=standard dialplan=0 localdialplan=4 cpndialplan=0 early_bconnect=no incoming_early_audio=no nodialtone=no presentation=-1 screen=-1 jitterbuffer=4000 jitterbuffer_upper_threshold=0 hdlc=no hold_allowed=yes [intern] ports=1 callgroup=1 pickupgroup=1 context=intern [extern] ports=2 context=extern msns=* echocancel=128 ___ --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] Remember last IP address of IAX client
Hello, does anybody know if it is possible to save the IP address of an IAX client logging into asterisk into the DB for future reference? I.e. one could distinguish between cases, where the client was last seen on the local net or on the road... even when it is not currently online. Cheers, Arik ___ --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] Re: Xen, Asterisk ISDN: Timing Problems
Thanks. What kernels do you use for dom0 and the domU's? Custom-built or out of the box? - Arik jason wrote: I would vote RAM. I've been using a FXO card in xen for a good year now with no issues at all. In fact, my zttest timings are the same between xen and native. Arik Raffael Funke wrote: Hi, is anybody running asterisk on a xen domU and can give an opinion on the following: I have delegated a FritzCard and a HFC card to my domU and installed an asterisk setup that was running on the same isdn hardware but on a dedicated machine flawlessly. I experienced what I believed to be timing problems: sometimes calls on the Fritzcard did not seem to reach asterisk, when calls were being made, sometimes they were horribly distorted. I quickly abandoned the project at the time for lack of time. I would now make another trial. Can anybody tell me if the problems I was having were more likely to result from the fact that the isdn hardware was dedicated to the domU (i.e. maybe that produces some sort of bottleneck!?) or from too little ram allocated to my domU? (I believe I had 128 MB or so) Thanks, Arik ___ --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] Re: Xen, Asterisk ISDN: Timing Problems
That has been fixed in the current Xen, and as far as I can tell works without problems. (At least for some NICs I had dedicated to another domU.) Regards, Arik Howard Lowndes wrote: I have to run Asterisk on the dom0 host as earlier versions of Xen had problems handing PCI control over to a domU kernel. Does anyone know if this has been fixed yet? Arik Raffael Funke wrote: Thanks. What kernels do you use for dom0 and the domU's? Custom-built or out of the box? - Arik jason wrote: I would vote RAM. I've been using a FXO card in xen for a good year now with no issues at all. In fact, my zttest timings are the same between xen and native. Arik Raffael Funke wrote: Hi, is anybody running asterisk on a xen domU and can give an opinion on the following: I have delegated a FritzCard and a HFC card to my domU and installed an asterisk setup that was running on the same isdn hardware but on a dedicated machine flawlessly. I experienced what I believed to be timing problems: sometimes calls on the Fritzcard did not seem to reach asterisk, when calls were being made, sometimes they were horribly distorted. I quickly abandoned the project at the time for lack of time. I would now make another trial. Can anybody tell me if the problems I was having were more likely to result from the fact that the isdn hardware was dedicated to the domU (i.e. maybe that produces some sort of bottleneck!?) or from too little ram allocated to my domU? (I believe I had 128 MB or so) Thanks, Arik ___ --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 ___ --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] Call file: CallerID problem
Hello,
I have the following call file:
Channel: Local/[EMAIL PROTECTED]/n
Callerid: 27
MaxRetries: 2
RetryTime: 10
Context: test2
Extension: s
And the following dialplan:
[test1]
exten = s,1,NoOp(${CALLERIDNUM})
But my CALLERIDNUM and CALLERIDNAME variables are both empty. I tried
without success following alternatives in the call file:
- Callerid: Someone 27
- SetVar: CALLERID(number)=27
Can anybody tell me, how to set the callerID for outgoing calls realised
with call files? I am using asterisk 1.2.
Thanks in advance for any hints.
Regards,
Arik
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[asterisk-users] Ring locally when home or roadwarrior via IAX when away
Hi, want to have calls directed to internal fixed phones, when my employees are home and automatically to their IAX connection when they are logged in remotely. How do I do this? The picture is as follows: --- HOME --- - user logged in via IAX with local IP adress - fixed line should ring - IAX should not ring --- ON THE ROAD --- - user logged in via IAX with an EXTERNAL IP - fixed line should not ring - but IAX should be called. I guess, in short my question reduces to: How do I find the IP adress of a specific iax client? Thanks in advance for the help. Regards, Arik ___ --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] Re: Ring locally when home or roadwarrior via IAX when away
Anselm Martin Hoffmeister wrote:
Am Montag, den 06.11.2006, 11:04 + schrieb Arik Raffael Funke:
I guess, in short my question reduces to: How do I find the IP adress of
a specific iax client?
With a set of statements like
exten = _5XX,1,Set(IPADDR=${CUT(${DB(SIP/sip${EXTEN})},:,1)})
exten = _5XX,2,GotoIf($[10.0. = ${IPADDR:0:5}]?100)
exten = _5XX,3,GotoIf($[ = ${IPADDR}]?100)
exten = _5XX,4,NoOp(Stuff for calling the client via SIP)
exten = _5XX,100,NoOp(Stuff for calling via the phoneline, cause user
is local)
This is untested though, but I hope to give you a useful hint here.
Thanks Anselm. Your solution does work for iax as well, except for some
typos. I have it working with following set of commands now, in case
anybody is wondering:
exten = s,1,Set(IPADDR=${CUT(DB(IAX/Registry/arik)|:|1)})
exten = s,2,GotoIf($[192.168.0. = ${IPADDR:0:10}]?100)
exten = s,3,GotoIf($[ = ${IPADDR}]?100)
exten = s,4,Dial(${ARIK_WEB},,t)
exten = s,100,Dial(${ARIK},,t)
Note ${ARIK} contains the fixed line channel, ${ARIK_WEB} the IAX
channel. arik is the iax login username.
Regards,
Arik
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[asterisk-users] chan_isdn / chan_sip problems
Hi, I am using Asterisk 1.2 with internal isdn phones connected via a hfcpci card in nt-mode with misdn. Bridging calls from the internal hfcpci via a avmfritz card (also chan_misdn) to the PSTN works flawlessly. However when I use a sip channel to route the outgoing call via voipstunt, it always rings three times and then gives me a busy indication. With my previous configuration, asterisk 1.0.10, zaphfc, chan_capi-cm this was no problem. I thought it was a sip problem and used sip debug but at the moment when the ringing switches to busy no debug messages appear. I also tried a softphone - it works fine with the same config. So I figure that it has something to do with the chan_misdn to chan_sip bridging. Below it the chan_misdn debug trace from the console at the moment when the switch from ringing to busy occurs. Does this tell anybody something that might help with my problem? Do I have a mistake in my misdn configuration? Thanks in advance for any hints. Best regards, Arik console debug trace - hestia*CLI hestia*CLI hestia*CLI P[ 1] *I IND :TIMEOUT oad:23 dad:070712976872 pid:21 state:DIALING P[ 1] -- state: DIALING P[ 1] I SEND:DISCONNECT oad:23 dad:070712976872 pid:21 P[ 1] -- bc_state:BCHAN_ACTIVATED P[ 1] *ec_disable P[ 1] I IND :RELEASE oad: dad: pid:21 state:DIALING P[ 1] hangup_chan P[ 1] - queue_hangup P[ 1] release_chan: bc with l3id: 10042 P[ 1] * RELEASING CHANNEL pid:21 ctx:macro-tsblcr dad:sip oad:23 state: DIALIN G P[ 1] I SEND:RELEASE_COMPLETE oad: dad: pid:21 P[ 1] -- bc_state:BCHAN_CLEANED Scheduling destruction of call '[EMAIL PROTECTED]' in 32000 ms Reliably Transmitting (no NAT) to 80.239.235.200:5060: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK2079ec6b;rport From: Arik sip:[EMAIL PROTECTED];tag=as7a95fade To: sip:[EMAIL PROTECTED] Destroying call '[EMAIL PROTECTED]' 12 headers, 0 lines CReliably Transmitting (no NAT) to 80.239.235.200:5060: OPTIONS sip:sip.voipstunt.com SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK65df4ea2;rport From: asterisk sip:[EMAIL PROTECTED];tag=as439face3 To: sip:sip.voipstunt.com Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX.235.200:5060: Max-Forwards: 70 Date: Fri, 22 Sep 2006 15:50:43 GMTbranch=z9hG4bK2079ec6b;rport Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] ---q: 102 CANCEL hestia*CLI -- SIP read from 80.239.235.200:5060: PTIONS SIP/2.0 200 Ok: 0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK65df4ea2;rport From: asterisk sip:[EMAIL PROTECTED];tag=as439face3 To: sip:sip.voipstunt.com Contact: sip:80.239.235.200:5060 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Supported: User-Agent: (Very nice Sip Registrar Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: ___ --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] mISDN problem: no version for capi_cmd2str found
Hi, has anybody gotten the following message when trying to modprobe a isdn adapter card driver? mISDN_capi: no version for capi_cmd2str found: kernel tainted. Anybody knows whether it's a problem or not and how to get rid of it? Cheers, Arik ___ --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] Dealing with FINAREA redirects
Hi, does anybody currently use voipstunt from finarea? I place a call to sip.voipstunt.com I get a 302 redirection. Unfortunately the second server seems to support only a different set of codecs than the first: -- Called [EMAIL PROTECTED] -- Got SIP response 302 Moved temporarely back from 194.120.0.203 -- Now forwarding mISDN/1-1 to 'SIP/[EMAIL PROTECTED]:5060' (thanks to SIP/voipstunt-081c1ba0) Sep 14 15:36:56 WARNING[12025]: chan_sip.c:2561 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) My question: How to I get asterisk to re-negotiate the codecs with the new handler? - Or am I interpreting something wrong here? Regards, Arik ___ --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] Virtualise asterisk on Xen
Hi, has anybody experience running asterisk on a (i.e. fedora-based) Xen system? What about mISDN support etc.? For a low-load system I thought about using: 1. Sempron 2800+ 2. some memory, in your opinion how much should I attribute to the asterisk guest system? 3. A AVM Fritz!PCI card for PSTN access 4. HFCPCI-S card in nt-mode for internal ISDN bus provision 5. Asterisk 1.2 with chan_misdn for the ISDN-card support It would be great to hear some of your thoughts on this set-up? Regards, Arik NB: I have the impression that virtualisation is not a big issue on this mailing list... Is that due to a show-stopper I overlooked, just because everything goes so smoothly that nobody even bothers to mention it ;-), or because everybody has plenty of hardware they can dedicate to their PBXs? ___ --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] I am not getting 302 redirects...
Hi, How do the 302 redirects work in asterisk, and what is the promiscredir directive doing? I am not getting the documentation on this. I have following happening on my asterisk box: -- Executing Dial(mISDN/1-1, SIP/[EMAIL PROTECTED]||Tt) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 302 Moved temporarely back from 194.120.0.202 -- Now forwarding mISDN/1-1 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/voipstunt-081de188) Sep 11 11:57:33 NOTICE[16463]: chan_local.c:498 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel Sep 11 11:57:33 NOTICE[16463]: app_dial.c:474 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]' (cause = 0) == Everyone is busy/congested at this time (1:0/0/1) - Can anybody tell me what the point is of asterisk trying out Local/[EMAIL PROTECTED] instead of the extension provided by the 302? Thanks in advance, Arik ___ --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] bristuff compile problems with kernel 2.6.17.11
Hi, has anybody had success compiling bristuff with kernel 2.6.17.11? Error messages are below... Cheers, Arik --- /usr/src/asterisk_1.0.10/bristuff-0.2.0-RC8s/zaptel-1.0.10/zaptel.c:6520: warning: passing argument 4 of 'class_device_create' from incompatible pointer type /usr/src/asterisk_1.0.10/bristuff-0.2.0-RC8s/zaptel-1.0.10/zaptel.c:6520: error: too few arguments to function 'class_device_create' make[2]: *** [/usr/src/asterisk_1.0.10/bristuff-0.2.0-RC8s/zaptel-1.0.10/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/asterisk_1.0.10/bristuff-0.2.0-RC8s/zaptel-1.0.10] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.17.11' make: *** [linux26] Error 2 ___ --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] Re: bristuff compile problems with kernel 2.6.17.11
Tzafrir Cohen wrote: On Thu, Sep 07, 2006 at 01:27:36PM +0200, Arik Raffael Funke wrote: Hi, has anybody had success compiling bristuff with kernel 2.6.17.11? Error messages are below... This is not code that is touched by the bristuff patch. Anyway, I'd try the latest 0.3.0 bristuff patch. Where is the code from then? I thought zaptel was part of bristuff. Anyway, has anybody been able to get it to work with kernel-2.6.17.11? I am not using bristuff 0.3.0 because it requires asterisk 1.2. All my configuration currently bases on 1.0 and I am not quite ready to move it to 1.2. Cheers, Arik ___ --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] Re: bristuff compile problems with kernel 2.6.17.11
Tzafrir Cohen wrote: On Thu, Sep 07, 2006 at 08:56:54PM +0200, Arik Raffael Funke wrote: Tzafrir Cohen wrote: On Thu, Sep 07, 2006 at 01:27:36PM +0200, Arik Raffael Funke wrote: Hi, has anybody had success compiling bristuff with kernel 2.6.17.11? Error messages are below... This is not code that is touched by the bristuff patch. Anyway, I'd try the latest 0.3.0 bristuff patch. Where is the code from then? I thought zaptel was part of bristuff. Anyway, has anybody been able to get it to work with kernel-2.6.17.11? ... snipp ... I am not using bristuff 0.3.0 because it requires asterisk 1.2. All my configuration currently bases on 1.0 and I am not quite ready to move it to 1.2. You can use zaptel 1.2 with Asterisk 1.0 . Though you'll have to do the patching yourself. We have tested this combination with analog zaptel lines, but not much so with BRI, though. Hi Tzafrir, thanks for the extensive reply. I tested bristuff-0.3.0 and its zaptel 1.2 compiles fine. - As you already said. If I wanted to use zaptel 1.2 with asterisk 1.0.10, do you think that the following would work: 1. download patch bristuff-0.3.0 contents with ./download.sh 2. do a make make install in the zaptel-1.2 directory 3. go back to bristuff 0.2.X and build install asterisk? Cheers, Arik ___ --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] Re: No ring tone on outgoing calls
Tim Sharp wrote: I am running on 1.2.7.1 and have an intermittent problem when making outgoing calls. Sometimes the calling party does not hear the ring tone in their handset, but the call goes through. From my extension I have only had 3 calls like this in the last couple of weeks, other people have had 3 or 4 calls in a single day and then not have a problem for a couple of days. The called phone number is not the problem because sometimes it works and sometimes not. We have both Aastra and Cisco phone sets and the problem occurs on both of them. We have SIP to PRI connections. I believe that this problem started after we upgraded from 1.0.9 but not 100 percent sure of that. Any help or suggestions that you have would be appreciated. Thank you, Tim I have a similar problem with following setup: Idefix on external network - IAX in - Asterisk - SIP out to VoIP-Stunt - Landline Also, the first few seconds of the conversation are missing. I.e. the other party answers the phone, but I never hear it. Audio goes through however after a bit. With following config I have no problem what-so-ever: ISDN phone - ZapHFC - Asterisk - Sip out to VoIP-Stunt - Landline Thus the problem lies somehow with the IAX in. Has anybody seen this before? Cheers, Arik ___ --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] Why are sip-channels too lagged?
Hello, I am getting lots of messages as the ones attached below. Is this a problem anybody can explain. (My internet connection is NOT slow or instable... thus I don't get it.) Maybe does this result from incorrect registration? Cheers, Arik - sip.conf -- [general] qualify=no srvlookup=yes canreinvite=yes register = x:[EMAIL PROTECTED]/ [sipgate] type=friend language=de insecure=very ; otherwise I get authentication errors nat=yes username= fromuser= fromdomain=sipgate.de secret= host=sipgate.de qualify=yes context=sipgate Jun 7 16:20:32 NOTICE[3160]: Peer 'sipgate' is now REACHABLE! Jun 7 16:22:36 NOTICE[3160]: Peer 'sipgate' is now UNREACHABLE! Jun 7 16:22:46 NOTICE[3160]: Peer 'sipgate' is now REACHABLE! Jun 7 16:26:51 NOTICE[3160]: Peer 'sipgate' is now UNREACHABLE! Jun 7 16:27:02 NOTICE[3160]: Peer 'sipgate' is now REACHABLE! Jun 7 16:29:06 NOTICE[3160]: Peer 'sipgate' is now UNREACHABLE! Jun 7 16:29:17 NOTICE[3160]: Peer 'sipgate' is now REACHABLE! Jun 7 16:29:31 WARNING[3160]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 12552 (Critical Request) Jun 7 16:29:45 NOTICE[3160]:-- Registration for '[EMAIL PROTECTED]' timed out, trying again Jun 7 16:31:22 NOTICE[3160]: Peer 'sipgate' is now UNREACHABLE! Jun 7 16:31:39 WARNING[3160]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 12554 (Critical Request) Jun 7 16:31:53 NOTICE[3160]:-- Registration for '[EMAIL PROTECTED]' timed out, trying again ___ --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
