[asterisk-users] Is answer() necessary ?
Hello list, is it necessary to properly answer() an incoming call ? I don't want to answer a call because the caller has to pay even if the attached SIP-phones do not answer the phone call. Because I answer() the incoming call, the caller has to pay for 60 seconds of 'ringtone'. On the other hand, sometimes an incoming call is send to a macro where the caller is given the opportunity to leave a voicemail message. It's to late to answer() the call in the macro, but I guess the voicemail()-application automatically anwers the call ?? How about an IVR-prompt and a queue ? Do I need to answer the incoming call before playing a voiceprompt and before sending it into a queue ?? Greetingz, Jonas. -- _ -- 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] Premicell solutions?
On 28 February 2010 15:28, Gordon Henderson gordon+aster...@drogon.net wrote: On Sun, 28 Feb 2010, LATEEF, IRFAN (ATTSI) wrote: Gordon , Are you referring to Femto Cells ?? No - devices like the Portech boxes - they take SIM card(s) and present each one as a SIP interface. Although I guess if you've got spare ISDN ports and are happy configuring them, then a GSM - ISDN device will work, but I think using SIP over Ethernet might be easier to get going (and possibly cheaper) e.g. any of the boxes on this page http://www.voipon.co.uk/portech-voip-gsm-gateway-c-3_192_193.html but obviously sourced local to whatever country you're in. Thank you! I am in the UK, so that page looks good. Are these devices something you have experience of using with Asterisk? I was only suggesting ISDN because I've heard of SIP interop issues, but not ISDN interop issues :) Regards, Steve -- _ -- 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] Is answer() necessary ?
You only need to answer() the call when you want to play audio, or music on hold, receive dtmf, etc. If you are just sending the incoming call to a Dial() or Queue(without music on hold), you don't need to answer. The receiving party will do the answering. This way the callee doesn't need to pay for the 'ringing'. If the call is coming in on a DAHDI connection, and you have a special agreement with your telco, you can do something called early audio. To do this, you need to do Progress() call, and then remember to tell the asterisk functions you are using, that they should not automatically answer the call. (Playback etc does this unless you explicitly tell it not to) If you are connected to your telco via SIP, i'm not sure how this works. I don't have any experience there. But I think it's about the same. About IVR-prompt and a queue.. Very few telcos, if any, let you receive DTMF in early audio. I'm not sure if it is supported in asterisk either. So I would say you always need to answer you call if you are going to have IVR-prompts before a queue. Asterisk does this for you anyways unless you tell it not to, when using functions like Background and Playback. Regards, Håkon On Mon, Mar 1, 2010 at 11:22 AM, jonas kellens jonas.kell...@telenet.be wrote: Hello list, is it necessary to properly answer() an incoming call ? I don't want to answer a call because the caller has to pay even if the attached SIP-phones do not answer the phone call. Because I answer() the incoming call, the caller has to pay for 60 seconds of 'ringtone'. On the other hand, sometimes an incoming call is send to a macro where the caller is given the opportunity to leave a voicemail message. It's to late to answer() the call in the macro, but I guess the voicemail()-application automatically anwers the call ?? How about an IVR-prompt and a queue ? Do I need to answer the incoming call before playing a voiceprompt and before sending it into a queue ?? Greetingz, Jonas. -- _ -- 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 -- _ -- 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] Is answer() necessary ?
jonas kellens wrote: Hello list, is it necessary to properly answer() an incoming call ? I don't want to answer a call because the caller has to pay even if the attached SIP-phones do not answer the phone call. Because I answer() the incoming call, the caller has to pay for 60 seconds of 'ringtone'. On the other hand, sometimes an incoming call is send to a macro where the caller is given the opportunity to leave a voicemail message. It's to late to answer() the call in the macro, but I guess the voicemail()-application automatically anwers the call ?? How about an IVR-prompt and a queue ? Do I need to answer the incoming call before playing a voiceprompt and before sending it into a queue ?? Greetingz, Jonas. Hi I was having similar problems to you and stoped using Answer at the start of my incoming dial plans with no problems. I'm finding that applications such as VoiceMail implicitly answer the channel so that is not a problem. Here's a more in depth response I received when asking the same question on this list ___ Recall that in regards to SIP implementation, Asterisk is a back-to-back user agent (B2BUA). This means that one logical call leg comes in, and another logical call leg is generated out, and the two are cross-connected. If SIP is not the signaling technology used on one or both channels, the effect is analogical where applicable. However, I will use SIP to illustrate the point; you can extrapolate from there similar effects on other channel types. The function that Answer() has on a signaling level is to effect an pickup on the incoming call leg. In SIP, this is a 200 OK message. If you then proceed to Dial() out on another channel, any ringback generated out the first channel will be in-band; that is to say, it will be inside the acoustic bearer. A far-end pickup (200 OK) is necessary to exchange audio bidirectionally. Some dial plan functions - mostly those that conceivably entail a two-way communication path - imply Answer() and will execute it for you if you have not already done so. Others do not. For example, it is possible to generate in-band ringback via early media, e.g. by sending a 183 Session in Progress message with an SDP payload to the sender. So, for example, if you were to do this: exten = s,1,MusicOnHold without doing an Answer() first, the MOH would be played via early media without pickup. By the same token, if you Dial() out before Answer()ing, the ringback generated will also be via early media (or, if applicable, out-of-band, depending on other settings): exten = s,1,Dial(SIP/otherpl...@other_peer) This will not result in a 200 OK received on the far end of the incoming channel until there is a 200 OK received on the near end of the outgoing channel. That is the function that Answer() serves. The option to remove it is contingent upon refraining from use of dial plan applications that implicitly invoke it. Alex Hope all this helps Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Premicell solutions?
On Mon, 1 Mar 2010, Steve Davies wrote: On 28 February 2010 15:28, Gordon Henderson gordon+aster...@drogon.net wrote: On Sun, 28 Feb 2010, LATEEF, IRFAN (ATTSI) wrote: Gordon , Are you referring to Femto Cells ?? No - devices like the Portech boxes - they take SIM card(s) and present each one as a SIP interface. Although I guess if you've got spare ISDN ports and are happy configuring them, then a GSM - ISDN device will work, but I think using SIP over Ethernet might be easier to get going (and possibly cheaper) e.g. any of the boxes on this page http://www.voipon.co.uk/portech-voip-gsm-gateway-c-3_192_193.html but obviously sourced local to whatever country you're in. Thank you! I am in the UK, so that page looks good. Are these devices something you have experience of using with Asterisk? I was only suggesting ISDN because I've heard of SIP interop issues, but not ISDN interop issues :) I've only used those for outgoing calls - when they seem to work just fine. I can't imagine there would be any issues with incoming - presumably get them to call some dialplan code to some sort of validation (PIN) then DISA to allow dialling an internal extension.. Gordon-- _ -- 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] MeetMe and usernum
hi, I am trying to get the usernum of a user when dialing in to a MeetMe conference. Is there somehow a possibility to save the usernum of a MeetMe participant into a variable? Everything should be done through the DialPlan, no manager and no *cli. Thanks for your help, Emrah -- _ -- 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] yahoo
Hello. tried to use Asterisk with yahoo and I saw that there was a software called GTalk2VoIP V8 by RZ and UGIN, Tyumen, Russia. You have one? -- _ -- 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] Swift from eagi, problems with prosody rate
Hi, I'm trying to use Swift tts from eagi, my problem is when I send EXEC SWIFT *prosody rate*=\'.8\' Hello World\, this is a test\,/*prosody* |0|1 Would I use a scape character? Thanks -- _ -- 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] MeetMe and usernum
On Mon, 1 Mar 2010, Emrah wrote: I am trying to get the usernum of a user when dialing in to a MeetMe conference. Is there somehow a possibility to save the usernum of a MeetMe participant into a variable? Everything should be done through the DialPlan, no manager and no *cli. I use 1.2, but I found I had to call an AGI that connected back to Asterisk via AMI to execute meetme list and then parse the result in the AGI. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rtcachefriends qualify
[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'gerrie' Am I correct that when I turn on rtcachefriends in sip.conf, database-changes in my MySQL-DB will not be reflected untill a reload ?? Am I correct that when I turn off qualify in my realtime sip-database, I could be confronted with NAT-problems for SIP-peers that are behind a NAT-router ? Is this the choice I need to take ? Greetingz, Jonas -- _ -- 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] Asterisk and Cisco DTMF
Hi, I have encountered a DTMF issue. My scenario: Access carrier-sip Asterisk-1.4.25.1-sipCiscoGW-ISDN-TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Kind regards, Szabolcs Szasz -- _ -- 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] rtcachefriends qualify
Hi, I think so, maybe someone can help clarify this for me also. I have: rtcachefriends=yes rtautoclear=yes in sip.conf and was under the impression that this caches the settings from the database until a user unregisters. When they unregister the data is removed from the cache (rtautoclear). For me this was a nice compromise. This is from memory but I’m pretty sure I got this from the documentation online, if someone can confirm what I’m saying that would be sweet. Thanks. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: 01 March 2010 14:06 To: Asterisk Mailing Subject: [asterisk-users] rtcachefriends qualify [Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'gerrie' Am I correct that when I turn on rtcachefriends in sip.conf, database-changes in my MySQL-DB will not be reflected untill a reload ?? Am I correct that when I turn off qualify in my realtime sip-database, I could be confronted with NAT-problems for SIP-peers that are behind a NAT-router ? Is this the choice I need to take ? Greetingz, Jonas -- _ -- 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] SPA3102 Firmware Upgrade via TFTP fails
Hi everyone, I'm trying to set up a VOIP mass deployment. To do so, I want to generate a configuration xml fails. I read somewhere that I had to use : http://phone -or- device ip address/admin/spacfg.xml but it work with an upper firmware only. My Software Version is 3.3.6(GW) The last firmware versin is 5.1.10(GW) Upgrade Enable: is set to yes on the linksys SPA3102 web interface TFTP works because I tried to download and upload to the TFTP server and it worked fine I tried to upgrade firmware via tftp using this command : http://192.168.0.1/upgrade?tftp://192.168.0.2/spa.bin but upgrade fails :s syslog : Mar 1 10:17:04 georghy-desktop kernel: [ 878.063638] :03:00.0: eth3: Link is Up 100 Mbps Full Duplex, Flow Control: RX/TX Mar 1 10:17:04 georghy-desktop kernel: [ 878.063645] :03:00.0: eth3: 10/100 speed: disabling TSO Mar 1 10:17:33 192.168.0.1 SPA-3102 00:0e:08:ce:be:4e — Requesting upgrade tftp://192.168.0.2:69/spa.bin Mar 1 10:17:38 192.168.0.1 SPA-3102 00:0e:08:ce:be:4e — Upgrade failed: tftp_get failed Mar 1 10:17:38 georghy-desktop kernel: [ 913.473031] :03:00.0: eth3: Link is Down Mar 1 10:17:43 192.168.0.1 ** postupgrade handling (0) what should I do ? -- Cordialement, / Greetings, Georghy FUSCO -- _ -- 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] Attended transfer: transferring a call as soon as the destination starts ringing
Hi all! Ext A, B and C are SIP phones. Ext A receives a call from Ext B. Ext A wants to transfer the call to Ext C. Ext A puts the first call on hold, dials Ext C, then simply hangs up as soon as the call to Ext C starts *ringing*. In other words, Ext B wants to be sure Ext C is ringing (i.e. it is not busy or unavailable) but doesn't want to talk to him. Unfortunately, as soon as Ext A hears Ext C is ringing and hangs up or hits Transfer, the call is closed and a *new* call from Ext B to Ext C starts. This way, Ext C sees an unanswered call from Ext A, which is an unexpected behaviour. I played with directmedia and directrtpsetup, but no success so far. Any ideas, please? Thanks in advance. Alex -- _ -- 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 and Cisco DTMF
On Mon, Mar 1, 2010 at 9:25 AM, Szasz Szabolcs szasz.szabo...@gmail.com wrote: Hi, I have encountered a DTMF issue. My scenario: Access carrier-sip Asterisk-1.4.25.1-sipCiscoGW-ISDN-TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Possible? Sure. Also possible that Cisco is passing along the in-band, as well as converting the out-of-band to in-band, ergo two for one. You can also tune the DTMF on the Cisco to ignore or set parameters on DTMF. Refer to the IOS guide for the appropriate arguments. Even worse, it's possible that you have a lot of echo, and the DTMF is echo-y enough that it gets interpreted as two-for-one. Can you take asterisk out of the loop, terminate sip carrier straight into Cisco for testing? -- _ -- 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 and Cisco DTMF
On Mon, Mar 1, 2010 at 9:40 AM, David Backeberg dbackeb...@gmail.com wrote: On Mon, Mar 1, 2010 at 9:25 AM, Szasz Szabolcs szasz.szabo...@gmail.com wrote: Hi, I have encountered a DTMF issue. My scenario: Access carrier-sip Asterisk-1.4.25.1-sipCiscoGW-ISDN-TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Can you take asterisk out of the loop, terminate sip carrier straight into Cisco for testing? You could also make a really simple dialplan object to do some DTMF directly with a channel on the asterisk, to see if things work properly going just that far. -- _ -- 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] MeetMe and usernum
On Mon, Mar 1, 2010 at 6:42 AM, Emrah e...@ekanet.net wrote: I am trying to get the usernum of a user when dialing in to a MeetMe conference. Is there somehow a possibility to save the usernum of a MeetMe participant into a variable? Everything should be done through the DialPlan, no manager and no *cli. You don't say what version you're running. I second Steve's claim. Even with 1.6, I can't think of how to do what you want without resorting to AGI. Which is technically in the dialplan, but you're going to have to do extra work elsewhere. If you're using 1.6, you will enjoy knowing about 'meetme list concise', which you can then process with awk. If you absolutely don't want to do AGI, you could always modify meetme.c, recompile, and share your work with others. I think you'll find that harder than writing an AGI. -- _ -- 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] Asterisk and Cisco DTMF
Thank you David, I did an IVR on Asterisk wich reads for the caller the entered digits, it's working. I did traces for the same call on access side (where the dtmfs come with rfc2833 method) and the other interface where I send the dtmfs with sip info method to Cisco gateway. Both side seems to be OK, no duplicates. Kind regards, Szabolcs Szasz On Mon, Mar 1, 2010 at 9:40 AM, David Backeberg dbackeberg at gmail.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: * On Mon, Mar 1, 2010 at 9:25 AM, Szasz Szabolcs szasz.szabolcs at gmail.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: ** Hi, ** ** I have encountered a DTMF issue. My scenario: ** ** Access carrier-sip ** Asterisk-1.4.25.1-sipCiscoGW-ISDN-TDM Switch ** ** the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk ** forwards it with SIP INFO method to Cisco gateway, but on TDM switch every ** digit is duplicated. Is it possible that the carrier sends inband along with ** rfc2833? ** Can you take asterisk out of the loop, terminate sip carrier straight ** into Cisco for testing? * You could also make a really simple dialplan object to do some DTMF directly with a channel on the asterisk, to see if things work properly going just that far. -- _ -- 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] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? - Email found in subject
Hi again! I have excellent success with the tiny fcpci and chan_capi, which is also working great with capi4hylafax. See net-dialup/fcpci-0.1-r1 in gentoo (should not be difficult to use this on other distros, but I have never done so). Do not confuse this with the fritzcapi! I managed to install fcpci and it seems to run fine (capiinfo output). Unfortunately i cant compile chan_capi against my Asterisk trunk r240716. Neither the trunk/head nor the 1.1.4 Version compiles. All fail with the following output: srvpbx:/usr/src/chan-capi-HEAD# make [CC] chan_capi.c - chan_capi.o In file included from chan_capi.c:32: chan_capi.h:34:26: error: asterisk/rtp.h: Datei oder Verzeichnis nicht gefunden chan_capi.c: In function âlocal_queue_frameâ: chan_capi.c:803: error: invalid operands to binary == (have âunion anonymousâ and âintâ) chan_capi.c: In function âinterface_cleanupâ: chan_capi.c:1071: warning: implicit declaration of function âast_rtp_destroyâ chan_capi.c: In function âsend_progressâ: chan_capi.c:1165: error: incompatible types in assignment chan_capi.c: In function âclear_channel_fax_loopâ: chan_capi.c:2884: error: invalid operands to binary == (have âunion anonymousâ and âintâ) chan_capi.c: In function âcapidev_handle_did_digitsâ: chan_capi.c:3548: error: incompatible types in assignment chan_capi.c: In function âcapi_queue_cause_controlâ: chan_capi.c:3564: warning: missing braces around initializer chan_capi.c:3564: warning: (near initialization for âfr.subclassâ) chan_capi.c:3569: error: incompatible types in assignment chan_capi.c:3573: error: incompatible types in assignment chan_capi.c: In function âcapidev_handle_info_indicationâ: chan_capi.c:3876: error: incompatible types in assignment chan_capi.c:3886: error: incompatible types in assignment chan_capi.c: In function âhandle_facility_indication_dtmfâ: chan_capi.c:4138: error: incompatible types in assignment chan_capi.c:4149: error: incompatible types in assignment chan_capi.c: In function âcapidev_handle_data_b3_indicationâ: chan_capi.c:4292: error: incompatible types in assignment chan_capi.c:4294: error: incompatible types in assignment chan_capi.c: In function âcapi_signal_answerâ: chan_capi.c:4316: warning: missing braces around initializer chan_capi.c:4316: warning: (near initialization for âfr.subclassâ) chan_capi.c: In function âcapidev_handle_disconnect_indicationâ: chan_capi.c:4605: warning: missing braces around initializer chan_capi.c:4605: warning: (near initialization for âfr.subclassâ) chan_capi.c:4654: error: incompatible types in assignment chan_capi.c: In function âcapidev_handle_connection_confâ: chan_capi.c:5025: warning: missing braces around initializer chan_capi.c:5025: warning: (near initialization for âfr.subclassâ) chan_capi.c: At top level: chan_capi.c:7746: warning: initialization from incompatible pointer type chan_capi.c: In function âconf_interfaceâ: chan_capi.c:8153: warning: passing argument 2 of âast_parse_allow_disallowâ from incompatible pointer type chan_capi.c:8156: warning: passing argument 2 of âast_parse_allow_disallowâ from incompatible pointer type make: *** [chan_capi.o] Fehler 1 Please excuse the messed up german output, i have yet to discover how to set a debian box to german keyboard, everything else english. I am limited to Asterisk trunk r240716 because i want to evaluate the T.38 - T.30 gateway function, which can only be patched into this revision. In my opinion there are the following alternatives in order to get the Fritz card running with Asterisk: A) Get chan-capi to compile: Unfortunately my C knowledge seems insufficent for this. I have found out that there is noch rtp.h in the asterisk source dir, only a file named rtp_engine.h. Changing the include accordingly unfortunately only fixes the very first error. I can't make any sense out of the second error (error: invalid operands to binary == (have union anonymous and int)), as, in line 803, the left-hand operand of the == is no union, let alone a anonymous union. I have already posted this at the chan-capi-users ML, but with no answer so far. B) Use a binary chan_capi.so from elsewhere?! I run Debian 5.0.2 Kernel 2.6.26-2-686. Could i take a compiled chan_capi.so from a machine with a different Asterisk 1.6 Version? C) Find another way to use fcpci eith Asterisk. Is this even possible? With mISDN? ...? Comments or any other Ideas are very appreciated. Sincerly Daniel Leese P.s.: Philipp, many thanks fo your answers so far ;) -- _ -- 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] Swift from eagi, problems with prosody rate
I solve the problem, was a string formated problem. Thanks On Mon, Mar 1, 2010 at 10:05 AM, equis software equissoftw...@gmail.comwrote: Hi, I'm trying to use Swift tts from eagi, my problem is when I send EXEC SWIFT *prosody rate*=\'.8\' Hello World\, this is a test\,/* prosody*|0|1 Would I use a scape character? Thanks -- _ -- 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] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? - Email found in subject
Hi, it seems that the asterisk API here was changed again and chan_capi must be adapted to this. I will have a look. Armin On Mon, 1 Mar 2010, dle...@lstelcom.com wrote: Hi again! I have excellent success with the tiny fcpci and chan_capi, which is also working great with capi4hylafax. See net-dialup/fcpci-0.1-r1 in gentoo (should not be difficult to use this on other distros, but I have never done so). Do not confuse this with the fritzcapi! I managed to install fcpci and it seems to run fine (capiinfo output). Unfortunately i cant compile chan_capi against my Asterisk trunk r240716. Neither the trunk/head nor the 1.1.4 Version compiles. All fail with the following output: srvpbx:/usr/src/chan-capi-HEAD# make [CC] chan_capi.c - chan_capi.o In file included from chan_capi.c:32: chan_capi.h:34:26: error: asterisk/rtp.h: Datei oder Verzeichnis nicht gefunden chan_capi.c: In function âlocal_queue_frameâ: chan_capi.c:803: error: invalid operands to binary == (have âunion anonymousâ and âintâ) chan_capi.c: In function âinterface_cleanupâ: chan_capi.c:1071: warning: implicit declaration of function âast_rtp_destroyâ chan_capi.c: In function âsend_progressâ: chan_capi.c:1165: error: incompatible types in assignment chan_capi.c: In function âclear_channel_fax_loopâ: chan_capi.c:2884: error: invalid operands to binary == (have âunion anonymousâ and âintâ) chan_capi.c: In function âcapidev_handle_did_digitsâ: chan_capi.c:3548: error: incompatible types in assignment chan_capi.c: In function âcapi_queue_cause_controlâ: chan_capi.c:3564: warning: missing braces around initializer chan_capi.c:3564: warning: (near initialization for âfr.subclassâ) chan_capi.c:3569: error: incompatible types in assignment chan_capi.c:3573: error: incompatible types in assignment chan_capi.c: In function âcapidev_handle_info_indicationâ: chan_capi.c:3876: error: incompatible types in assignment chan_capi.c:3886: error: incompatible types in assignment chan_capi.c: In function âhandle_facility_indication_dtmfâ: chan_capi.c:4138: error: incompatible types in assignment chan_capi.c:4149: error: incompatible types in assignment chan_capi.c: In function âcapidev_handle_data_b3_indicationâ: chan_capi.c:4292: error: incompatible types in assignment chan_capi.c:4294: error: incompatible types in assignment chan_capi.c: In function âcapi_signal_answerâ: chan_capi.c:4316: warning: missing braces around initializer chan_capi.c:4316: warning: (near initialization for âfr.subclassâ) chan_capi.c: In function âcapidev_handle_disconnect_indicationâ: chan_capi.c:4605: warning: missing braces around initializer chan_capi.c:4605: warning: (near initialization for âfr.subclassâ) chan_capi.c:4654: error: incompatible types in assignment chan_capi.c: In function âcapidev_handle_connection_confâ: chan_capi.c:5025: warning: missing braces around initializer chan_capi.c:5025: warning: (near initialization for âfr.subclassâ) chan_capi.c: At top level: chan_capi.c:7746: warning: initialization from incompatible pointer type chan_capi.c: In function âconf_interfaceâ: chan_capi.c:8153: warning: passing argument 2 of âast_parse_allow_disallowâ from incompatible pointer type chan_capi.c:8156: warning: passing argument 2 of âast_parse_allow_disallowâ from incompatible pointer type make: *** [chan_capi.o] Fehler 1 Please excuse the messed up german output, i have yet to discover how to set a debian box to german keyboard, everything else english. I am limited to Asterisk trunk r240716 because i want to evaluate the T.38 - T.30 gateway function, which can only be patched into this revision. In my opinion there are the following alternatives in order to get the Fritz card running with Asterisk: A) Get chan-capi to compile: Unfortunately my C knowledge seems insufficent for this. I have found out that there is noch rtp.h in the asterisk source dir, only a file named rtp_engine.h. Changing the include accordingly unfortunately only fixes the very first error. I can't make any sense out of the second error (error: invalid operands to binary == (have union anonymous and int)), as, in line 803, the left-hand operand of the == is no union, let alone a anonymous union. I have already posted this at the chan-capi-users ML, but with no answer so far. B) Use a binary chan_capi.so from elsewhere?! I run Debian 5.0.2 Kernel 2.6.26-2-686. Could i take a compiled chan_capi.so from a machine with a different Asterisk 1.6 Version? C) Find another way to use fcpci eith Asterisk. Is this even possible? With mISDN? ...? Comments or any other Ideas are very appreciated. Sincerly Daniel Leese P.s.: Philipp, many thanks fo your answers so far ;) -- _ -- 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] Erika DeBenedictis-Recommendation
oops, sent to wrong address by mistake! Sorry! On Mon, Mar 1, 2010 at 9:00 AM, drew einhorn drew.einh...@gmail.com wrote: -- Forwarded message -- From: Celia Einhorn celia.einh...@gmail.com Date: Wed, Feb 17, 2010 at 8:15 PM Subject: Fwd: Erika DeBenedictis-Recommendation To: drew einhorn drew.einh...@gmail.com -- Forwarded message -- From: David H. Kratzer d...@lanl.gov Date: Tue, Feb 16, 2010 at 9:24 AM Subject: Fwd: Erika DeBenedictis-Recommendation To: Larry Cox lj...@lanl.gov Cc: Celia Einhorn celia.einh...@gmail.com, Betsy Frederick betsy.freder...@gmail.com I just now realized I had not added your email address to the Supercomputing Challenge Scholarship judges list. I think this is the first communication that address has had. We are going to put up a web form for the students to submit all of their information, but that isn't finished yet so some folks are emailing the information per last year's instructions. More to come, David Date: Sun, 14 Feb 2010 23:41:25 -0700 Subject: Erika DeBenedictis-Recommendation From: Chris Hong haochen.h...@gmail.com To: scholarshi...@challenge.nm.org To Whom it may concern: I'm writing this letter in support of Erika DeBenedictis in light of the Challenge's scholarship opportunities. First and foremost, in terms of competition, Erika always brings a top-notch, if not winning, project to the challenge every year. More than just her competitive contributions, however, I believe Erika truly embraces what Supercomputing stands for - willing, mutual collaboration in a team of research scientists. Being on a team with Erika, last year was my first year competing in the challenge. From the kickoff, I immediately noticed Erika's dedication and involvement with the challenge. Being the first among us to arrive and the last to leave, she helped set up the various workshops to even taught programming seminars to the younger competitors - all with complete enjoyment and satisfaction. On a more personal level, as an unfamiliar first time competitor, whenever frustrations or difficulties arose, Erika was a person I could turn to. Programming with two other people was exceedingly difficult in particular, and yet Erika's patience, experience, leadership, and willingness to compromise and collaborate saw our winning project through. From her, I learned more than just technical programming skills, but the social and interpersonal skills necessary to succeed and integrate in a scientific team. In this sense, I owe her much. As both an ardent competitor and a genuinely concerned supporter of the challenge, Erika is deserving of your scholarship that could aid in extending her knowledge and scientific passions at the collegiate level - I honestly cannot imagine anyone else more fitting. Sincerely, Chris Hong -- Celia Bedelia Computer Fairy -- Drew Einhorn -- Drew Einhorn -- _ -- Bandwidth and Colocation 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] MeetMe and usernum
Hi! Thanks a lot for your answer. The problem with the command you mentioned is... When do I call it? If two people happen to enter the conf at the sametime, I have a feeling there may be some little confusion there... Do you think I could use the agi-background option with meetme? I am using 1.6. Thanks again guys! Emrah On Mon, Mar 01, 2010 at 09:45:33AM -0500, David Backeberg dbackeb...@gmail.com wrote: On Mon, Mar 1, 2010 at 6:42 AM, Emrah e...@ekanet.net wrote: I am trying to get the usernum of a user when dialing in to a MeetMe conference. Is there somehow a possibility to save the usernum of a MeetMe participant into a variable? Everything should be done through the DialPlan, no manager and no *cli. You don't say what version you're running. I second Steve's claim. Even with 1.6, I can't think of how to do what you want without resorting to AGI. Which is technically in the dialplan, but you're going to have to do extra work elsewhere. If you're using 1.6, you will enjoy knowing about 'meetme list concise', which you can then process with awk. If you absolutely don't want to do AGI, you could always modify meetme.c, recompile, and share your work with others. I think you'll find that harder than writing an AGI. -- _ -- 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 -- Emrah KAVUN e...@ekanet.net -- _ -- 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] Fwd: Erika DeBenedictis-Recommendation
-- Forwarded message -- From: Celia Einhorn celia.einh...@gmail.com Date: Wed, Feb 17, 2010 at 8:15 PM Subject: Fwd: Erika DeBenedictis-Recommendation To: drew einhorn drew.einh...@gmail.com -- Forwarded message -- From: David H. Kratzer d...@lanl.gov Date: Tue, Feb 16, 2010 at 9:24 AM Subject: Fwd: Erika DeBenedictis-Recommendation To: Larry Cox lj...@lanl.gov Cc: Celia Einhorn celia.einh...@gmail.com, Betsy Frederick betsy.freder...@gmail.com I just now realized I had not added your email address to the Supercomputing Challenge Scholarship judges list. I think this is the first communication that address has had. We are going to put up a web form for the students to submit all of their information, but that isn't finished yet so some folks are emailing the information per last year's instructions. More to come, David Date: Sun, 14 Feb 2010 23:41:25 -0700 Subject: Erika DeBenedictis-Recommendation From: Chris Hong haochen.h...@gmail.com To: scholarshi...@challenge.nm.org To Whom it may concern: I'm writing this letter in support of Erika DeBenedictis in light of the Challenge's scholarship opportunities. First and foremost, in terms of competition, Erika always brings a top-notch, if not winning, project to the challenge every year. More than just her competitive contributions, however, I believe Erika truly embraces what Supercomputing stands for - willing, mutual collaboration in a team of research scientists. Being on a team with Erika, last year was my first year competing in the challenge. From the kickoff, I immediately noticed Erika's dedication and involvement with the challenge. Being the first among us to arrive and the last to leave, she helped set up the various workshops to even taught programming seminars to the younger competitors - all with complete enjoyment and satisfaction. On a more personal level, as an unfamiliar first time competitor, whenever frustrations or difficulties arose, Erika was a person I could turn to. Programming with two other people was exceedingly difficult in particular, and yet Erika's patience, experience, leadership, and willingness to compromise and collaborate saw our winning project through. From her, I learned more than just technical programming skills, but the social and interpersonal skills necessary to succeed and integrate in a scientific team. In this sense, I owe her much. As both an ardent competitor and a genuinely concerned supporter of the challenge, Erika is deserving of your scholarship that could aid in extending her knowledge and scientific passions at the collegiate level - I honestly cannot imagine anyone else more fitting. Sincerely, Chris Hong -- Celia Bedelia Computer Fairy -- Drew Einhorn -- _ -- 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] MeetMe and usernum
On Mon, Mar 1, 2010 at 11:01 AM, Emrah e...@ekanet.net wrote: Hi! Thanks a lot for your answer. The problem with the command you mentioned is... When do I call it? If two people happen to enter the conf at the sametime, I have a feeling there may be some little confusion there... Do you think I could use the agi-background option with meetme? I am using 1.6. You'll need to figure out the channel the caller was originally on before you dump them into meetme, then grep for that channel on the output of meetme list to figure out their number in the meetme room. I personally would fire up an agi, pick off the channel, put them in the room, then grep on the meetme list, then set / store the variable. -- _ -- Bandwidth and Colocation 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] Unable to register a sip account with x-lite
Hi, I'm running Asterisk 1.6.2.2 on a linux red hat machine. I'm attempting to connect to asterisk using x-lite, but always get the '401 unauthorized' error back from the server. As far as I know I have my sip.conf and extensions.conf configured correctly and I've tried changing lots of things but am still getting nowhere. In my sip.conf I have: [user1] type=friend secret=user1 callerid=user1 qualify=yes nat=yes host=dynamic canreinvite=no context=internal disallow=all allow=gsm allow=ulaw allow=alaw Note I've tried nat=no and qualify=no but it doesn't change anything. Extensions.conf has: [internal] exten = user1,1,Dial(SIP/user1,10,r) exten = 1359,1,Dial(SIP/user1,10,r) exten = user1,hint,SIP/user1 If I could even get some debug from Asterisk showing me why the user is unauthorized that would be great. I have added verbose and debug to the messages line in logger.conf and also set 'sip set debug on'. But I can't find any debug output anywhere. The messages file just contains start up messages for asterisk and no debug output. I can't find any sip debug output anywhere. Anybody know what I can do to try and isolate my configuration problems? How can I get debug trace from Asterisk showing the connection coming in from x-lite? Many thanks, Tim - Tim Culhane, Critical Path Ireland, 42-47 Lower Mount Street, Dublin 2. Direct line: 353-1-2415107 phone: 353-1-2415000 tim.culh...@criticalpath.net http://www.criticalpath.net Critical Path a global leader in digital communications -- _ -- Bandwidth and Colocation 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] Unable to register a sip account with x-lite
Take the quotes off of user1 and restart asterisk. Do core set verbose 10 and core set debug 10, then try to connect your x-lite phone. If you still get nothing, change the ICE settings on the x-lite. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Culhane Sent: Monday, March 01, 2010 10:39 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unable to register a sip account with x-lite Hi, I'm running Asterisk 1.6.2.2 on a linux red hat machine. I'm attempting to connect to asterisk using x-lite, but always get the '401 unauthorized' error back from the server. As far as I know I have my sip.conf and extensions.conf configured correctly and I've tried changing lots of things but am still getting nowhere. In my sip.conf I have: [user1] type=friend secret=user1 callerid=user1 qualify=yes nat=yes host=dynamic canreinvite=no context=internal disallow=all allow=gsm allow=ulaw allow=alaw Note I've tried nat=no and qualify=no but it doesn't change anything. Extensions.conf has: [internal] exten = user1,1,Dial(SIP/user1,10,r) exten = 1359,1,Dial(SIP/user1,10,r) exten = user1,hint,SIP/user1 If I could even get some debug from Asterisk showing me why the user is unauthorized that would be great. I have added verbose and debug to the messages line in logger.conf and also set 'sip set debug on'. But I can't find any debug output anywhere. The messages file just contains start up messages for asterisk and no debug output. I can't find any sip debug output anywhere. Anybody know what I can do to try and isolate my configuration problems? How can I get debug trace from Asterisk showing the connection coming in from x-lite? Many thanks, Tim - Tim Culhane, Critical Path Ireland, 42-47 Lower Mount Street, Dublin 2. Direct line: 353-1-2415107 phone: 353-1-2415000 tim.culh...@criticalpath.net http://www.criticalpath.net Critical Path a global leader in digital communications -- _ -- Bandwidth and Colocation 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] Unable to register a sip account with x-lite
Hi,
Tim Culhane wrote:
Hi,
I'm running Asterisk 1.6.2.2 on a linux red hat machine.
I'm attempting to connect to asterisk using x-lite, but always get the
'401 unauthorized' error back from the server.
As far as I know I have my sip.conf and extensions.conf configured
correctly and I've tried changing lots of things but am still getting
nowhere.
In my sip.conf I have:
[user1]
type=friend
secret=user1
callerid=user1
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=internal
disallow=all
allow=gsm
allow=ulaw
allow=alaw
This got no issues.
Note I've tried nat=no and qualify=no but it doesn't change anything.
Extensions.conf has:
[internal]
exten = user1,1,Dial(SIP/user1,10,r)
exten = 1359,1,Dial(SIP/user1,10,r)
exten = user1,hint,SIP/user1
If I could even get some debug from Asterisk showing me why the user is
unauthorized that would be great. I have added verbose and debug to the
messages line in logger.conf and also set 'sip set debug on'. But I
can't find any debug output anywhere. The messages file just contains
start up messages for asterisk and no debug output. I can't find any sip
debug output anywhere.
Anybody know what I can do to try and isolate my configuration problems?
How can I get debug trace from Asterisk showing the connection coming in
from x-lite?
As for x-lite, you should use user1 in User name, Password and
Authorization user name
Many thanks,
Tim
-
Tim Culhane,
Critical Path Ireland,
42-47 Lower Mount Street,
Dublin 2.
Direct line: 353-1-2415107
phone: 353-1-2415000
tim.culh...@criticalpath.net
http://www.criticalpath.net
Critical Path
a global leader in digital communications
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[asterisk-users] Asterisk / Trixbox 2.6 Streaming MOH Problems
I've tried a number of solutions, but I've been unable to get Asterisk working with streaming MOH without running into the buffer issue. I've tried using various combinations madplay, mpg123, mpg321. I've also tried streamplayer by itself, and in combination with play-fifo ( http://www.freeswitch.org/asterisk_stuff/play-fifo.c ) to try and eliminate the issue. For those that are unaware of the problem, what happens when you use a streaming music source with asterisk is you have a process that is running all the time that pipes MOH into stdout, which is then read by asterisk. When a caller is on hold, asterisk starts reading from stdin, and you get your music on hold. When the caller hangs up, asterisk stops reading from stdin (and the pipe becomes blocking), and a buffer is created (I'm not sure where the buffer resides, although I suspect it's probably the system fifo pipe buffer). The problem becomes, when the next caller comes in, and is put on hold, you will hear that buffer (usually about 20-30 seconds), and then it will jump to the current position in the stream, so you hear an ugly jump between the middle of two songs. There was a magic version of mpg123 that was supposed to solve this problem (0.59r, I believe), but I've been unable to get this to work. For those interested, I'm streaming music off of a Barix Instreamer, attached to a satellite radio source (and yes, I'm paying the proper licence fees). The only thing I've found that works so far is a pretty ugly (although ingenious) hack as seen here (http://www.mail-archive.com/asterisk-users@lists.digium.com/msg197299.html), which creates its own host of problems (such as not being able to do a restart when convienent since it generates a call on its own (that's always running), so it's never convienent for asterisk to restart. Also, when I do restart asterisk, I have to restart the call, so I'd prefer having to go this route if at all possible. Another solution would be if asterisk could spawn a new process for every MOH caller. Is this possible? Does anyone have a successful deployment of streaming music on hold that they'd care to share? I'm using Asterisk 1.4 as part of Trixbox 2.6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? - Email found in subject
Hi, chan_capi trunk should be compilable now with current asterisk trunk. Armin On Mon, 1 Mar 2010, Armin Schindler wrote: Hi, it seems that the asterisk API here was changed again and chan_capi must be adapted to this. I will have a look. Armin On Mon, 1 Mar 2010, dle...@lstelcom.com wrote: Hi again! I have excellent success with the tiny fcpci and chan_capi, which is also working great with capi4hylafax. See net-dialup/fcpci-0.1-r1 in gentoo (should not be difficult to use this on other distros, but I have never done so). Do not confuse this with the fritzcapi! I managed to install fcpci and it seems to run fine (capiinfo output). Unfortunately i cant compile chan_capi against my Asterisk trunk r240716. Neither the trunk/head nor the 1.1.4 Version compiles. All fail with the following output: srvpbx:/usr/src/chan-capi-HEAD# make [CC] chan_capi.c - chan_capi.o In file included from chan_capi.c:32: chan_capi.h:34:26: error: asterisk/rtp.h: Datei oder Verzeichnis nicht gefunden chan_capi.c: In function âlocal_queue_frameâ: chan_capi.c:803: error: invalid operands to binary == (have âunion anonymousâ and âintâ) chan_capi.c: In function âinterface_cleanupâ: chan_capi.c:1071: warning: implicit declaration of function âast_rtp_destroyâ chan_capi.c: In function âsend_progressâ: chan_capi.c:1165: error: incompatible types in assignment chan_capi.c: In function âclear_channel_fax_loopâ: chan_capi.c:2884: error: invalid operands to binary == (have âunion anonymousâ and âintâ) chan_capi.c: In function âcapidev_handle_did_digitsâ: chan_capi.c:3548: error: incompatible types in assignment chan_capi.c: In function âcapi_queue_cause_controlâ: chan_capi.c:3564: warning: missing braces around initializer chan_capi.c:3564: warning: (near initialization for âfr.subclassâ) chan_capi.c:3569: error: incompatible types in assignment chan_capi.c:3573: error: incompatible types in assignment chan_capi.c: In function âcapidev_handle_info_indicationâ: chan_capi.c:3876: error: incompatible types in assignment chan_capi.c:3886: error: incompatible types in assignment chan_capi.c: In function âhandle_facility_indication_dtmfâ: chan_capi.c:4138: error: incompatible types in assignment chan_capi.c:4149: error: incompatible types in assignment chan_capi.c: In function âcapidev_handle_data_b3_indicationâ: chan_capi.c:4292: error: incompatible types in assignment chan_capi.c:4294: error: incompatible types in assignment chan_capi.c: In function âcapi_signal_answerâ: chan_capi.c:4316: warning: missing braces around initializer chan_capi.c:4316: warning: (near initialization for âfr.subclassâ) chan_capi.c: In function âcapidev_handle_disconnect_indicationâ: chan_capi.c:4605: warning: missing braces around initializer chan_capi.c:4605: warning: (near initialization for âfr.subclassâ) chan_capi.c:4654: error: incompatible types in assignment chan_capi.c: In function âcapidev_handle_connection_confâ: chan_capi.c:5025: warning: missing braces around initializer chan_capi.c:5025: warning: (near initialization for âfr.subclassâ) chan_capi.c: At top level: chan_capi.c:7746: warning: initialization from incompatible pointer type chan_capi.c: In function âconf_interfaceâ: chan_capi.c:8153: warning: passing argument 2 of âast_parse_allow_disallowâ from incompatible pointer type chan_capi.c:8156: warning: passing argument 2 of âast_parse_allow_disallowâ from incompatible pointer type make: *** [chan_capi.o] Fehler 1 Please excuse the messed up german output, i have yet to discover how to set a debian box to german keyboard, everything else english. I am limited to Asterisk trunk r240716 because i want to evaluate the T.38 - T.30 gateway function, which can only be patched into this revision. In my opinion there are the following alternatives in order to get the Fritz card running with Asterisk: A) Get chan-capi to compile: Unfortunately my C knowledge seems insufficent for this. I have found out that there is noch rtp.h in the asterisk source dir, only a file named rtp_engine.h. Changing the include accordingly unfortunately only fixes the very first error. I can't make any sense out of the second error (error: invalid operands to binary == (have union anonymous and int)), as, in line 803, the left-hand operand of the == is no union, let alone a anonymous union. I have already posted this at the chan-capi-users ML, but with no answer so far. B) Use a binary chan_capi.so from elsewhere?! I run Debian 5.0.2 Kernel 2.6.26-2-686. Could i take a compiled chan_capi.so from a machine with a different Asterisk 1.6 Version? C) Find another way to use fcpci eith Asterisk. Is this even possible? With mISDN? ...? Comments or any other Ideas are very appreciated. Sincerly Daniel Leese P.s.: Philipp, many thanks fo your answers so far ;) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To
Re: [asterisk-users] MeetMe and usernum
On Mon, 1 Mar 2010, Emrah wrote:
The problem with the command you mentioned is... When do I call it? If
two people happen to enter the conf at the sametime, I have a feeling
there may be some little confusion there...
If you are using a database, it may provide generic locking that you can
[ab]use.
For example, MySQL provides get_lock() which could be used like:
select get_lock('find-usernum', 20);
This would block another process from executing until:
1) 20 seconds is up.
2) the locking process exits.
3) you explicitly release the lock by executing release_lock().
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Server response time
On 2/28/2010 10:21 AM, Gordon Henderson wrote: On Sun, 28 Feb 2010, Juan C. Villa wrote: Hey Guys, I am considering leasing a new server in Germany to run my Asterisk infrastructure and I was wondering how response time would affect the performance of the system. Right now I have a response time of around 60-70ms with my server in California. The server in Germany would have a response time of around 140ms (both ways). My DID/Termination providers are in Canada and the USA, and all my voip boxes are also in the USA. Any suggestions or recommendations? Being based in the UK, I'd say why not the UK rather then Germany - we're closer to the US after-all :) However, one thing we don't know: Where are you and your customers based? I also find it odd that a lot of people UK based still think they can get better deals (cheaper more b/w) by hosting in the US rather than in the UK - so I'm curious as to why you'd want to host outside the US... But as long as you're not passing media then anywhere you have good connectivity ought to work - however if you are passing media, then I'd be concerned that someone in California is calling their neighbour and the data is going all the way to Germany and back again... That really will be noticeable... Gordon In response to Gordon: Hetzner offers the best dedicated server deal I have every seen. I have been a Cari.net client for over a year now, but I am needing a more powerful server and I don't want to pay $200+ a month for it. Hetzner has a connection to the Level 3 network that recently installed a transoceanic fiber optic link with a lag of less than 40 ms. The total lag from Germany to USA (2 way) is around ~110ms (Just tested it today). Who this cause any issues with my VoIP applications? Right now I have two VoIP boxes installed in Switzerland which are connected to my server in California (avg response time = 190ms) and I have no problems at all. What would you guys advice? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No RTP from asterisk?
- Tri Tu mtr...@yahoo.com wrote: RTP is only firewall issue. Make sure that you can pass traffic from your client to the asterisk server. If it's on the same LAN, there shouldn't be any issue with RTP unless the Asterisk is setup with firewall to block RTP traffic (default is from 1 - 2 upd) Asterisk doesn't support G29 (pass-through is OK) but if you want to connect from your client to asterisk server with G729, you need to buy license. Using G711 is free and it taking about 68kbp. From: Peter Serwe peter.se...@gmail.com To: asterisk-users@lists.digium.com Sent: Sat, February 27, 2010 12:42:56 PM Subject: [asterisk-users] No RTP from asterisk? I've got an asterisk installation of 1.4.30-RC2 running, and while I can register lines and get call setup to pass, for some reason no RTP is being generated or received by asterisk. Debug doesn't seem to give me too much of relevance about it, especially rtp debug. I had a few other small issues, like trying to negotiate G729 when it's not capable, but since then, I've changed everything back to G711. I have connected to it, a SIP trunk, 3 registered users and I'm at a loss as to how to troubleshoot this further. Can anyone point me in the right direction? Peter -- Thanks, Phil http://issues.asterisk.org/view.php?id=16929 -- _ -- Bandwidth and Colocation 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] Server response time
Juan C. Villa a écrit : [...] The total lag from Germany to USA (2 way) is around ~110ms (Just tested it today). Who this cause any issues with my VoIP applications? Right now I have two VoIP boxes installed in Switzerland which are connected to my server in California (avg response time = 190ms) and I have no problems at all. What would you guys advice? FYI, I made an mtr to the IP 143.215.103.174, one from one of our servers in Switzerland, the second from an Hetzner one: both give 112 ms AVG time. Regards -- Daniel -- _ -- Bandwidth and Colocation 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] OT:4 Line DECT Cordless phone without answering machine
Anyone know of a cordless 4 line DECT phone that does't have an answering machine? Or one that costs less than $300.00 (even with answering)? Yes I tried Google but G too many results still sifting thru. TIA -- _ -- Bandwidth and Colocation 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] OT:4 Line DECT Cordless phone without answering machine
On Mon, 1 Mar 2010, C F wrote: Anyone know of a cordless 4 line DECT phone that does't have an answering machine? Or one that costs less than $300.00 (even with answering)? 4 line? Do you mean 4 handsets or 4 SIP accounts? Siemes gigaset range - the down-side is that they can only make 2 concurrent SIP calls on one base station. So 4 (actually 6) SIP accounts, up to 6 handsets but only 2 concurrent calls. Gordon -- _ -- Bandwidth and Colocation 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] Solved:Re: OT:4 Line DECT Cordless phone without answering machine
Base: http://www.amazon.com/RCA-25424RE1-4-Line-Expandable-Waiting/dp/tech-data/B000UVRYAI/ref=de_a_smtd Cordless handset: http://www.amazon.com/RCA-H5401RE1-Accessory-Handset-25423/dp/tech-data/B000UVQ6OI/ref=de_a_smtd Less than $170.00 for both :). thanks anyhow On Mon, Mar 1, 2010 at 5:08 PM, C F shma...@gmail.com wrote: Anyone know of a cordless 4 line DECT phone that does't have an answering machine? Or one that costs less than $300.00 (even with answering)? Yes I tried Google but G too many results still sifting thru. TIA -- _ -- Bandwidth and Colocation 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] OT:4 Line DECT Cordless phone without answering machine
On Mon, Mar 1, 2010 at 5:16 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Mon, 1 Mar 2010, C F wrote: Anyone know of a cordless 4 line DECT phone that does't have an answering machine? Or one that costs less than $300.00 (even with answering)? 4 line? Do you mean 4 handsets or 4 SIP accounts? Sorry should have specified, I meant 4 line analog. See my other post I already found one. Siemes gigaset range - the down-side is that they can only make 2 concurrent SIP calls on one base station. So 4 (actually 6) SIP accounts, up to 6 handsets but only 2 concurrent calls. Gordon -- _ -- Bandwidth and Colocation 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] User on PC?
I'm looking for a way for linux to query a pc if user X is on, and has used the pc recently or the screensaver is not active. If so, I'll route a call for user X to the phone near that PC. Ideas, anyone? Leif -- _ -- Bandwidth and Colocation 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] OT:4 Line DECT Cordless phone without answering machine
On Mon, 1 Mar 2010 17:26:26 -0500, C F wrote: On Mon, Mar 1, 2010 at 5:16 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Mon, 1 Mar 2010, C F wrote: Anyone know of a cordless 4 line DECT phone that does't have an answering machine? Or one that costs less than $300.00 (even with answering)? 4 line? Do you mean 4 handsets or 4 SIP accounts? Sorry should have specified, I meant 4 line analog. See my other post I already found one. Siemes gigaset range - the down-side is that they can only make 2 concurrent SIP calls on one base station. So 4 (actually 6) SIP accounts, up to 6 handsets but only 2 concurrent calls. The Panasonic KX-TG5000, or related follow-on models will do this. They have a largish desk phone that includes the DECT base, then support up to 8 cordless handsets. The base handles 4 analog lines and includes a built-in battery backup. Long ago I had the KX-TG4000, which was an early model in the series that operated on 2.4 GHz using proprietary coding over the radio links. A later model went to 5.8 GHz, then another finally used DECT 6.0. These are more costly than you describe. The base 1 handset usually selling for around $400, with extra handsets $100 each. They used to offer a cordless handset in the form factor of a nice little desk phone, not just the usual candy bar style. Michael Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation 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] User on PC?
On Mon, 1 Mar 2010, Leif Neland wrote: I'm looking for a way for linux to query a pc if user X is on, and has used the pc recently or the screensaver is not active. If so, I'll route a call for user X to the phone near that PC. Ideas, anyone? In the good old days we had utilities such as finger and so on - these days we're all paranoid about security, etc. that all the nice stuff is turned off... Or should be! However, if you're on a LAN/WAN at the company level then they can be made to work again, but outside a strictly controlled environment it's not going to be easy - unless you write something to do it yourself... Gordon -- _ -- Bandwidth and Colocation 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] help!!! Internal extensions not connect
I have a problem with my internal extensions, I'm using Asterisk 1.6.2.5 and freePBX 2.6. When I call betwen extensions these don't connect. There is a long silence and finally hang up. I have an E1 whit r2 and I use openr2, but I don't have problems to do calls to the PSTN..my problem it only with internal extension. Please help...it's an Asterisk bug??. Thanks :S -- Carem Gyssell Nieto Garcia -- _ -- Bandwidth and Colocation 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] User on PC?
On Mon, 2010-03-01 at 23:46 +0100, Leif Neland wrote: I'm looking for a way for linux to query a pc if user X is on, and has used the pc recently or the screensaver is not active. If so, I'll route a call for user X to the phone near that PC. If you're using a relatively modern version of Asterisk, you could use the res_jabber and the JABBER_STATUS function to see if they're marked as available in their XMPP IM client. (Most IM clients will set the status to away when the screensaver kicks in.) -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help!!! Internal extensions not connect
You doesn't seem to have a proper context,extension,priority available for internal calls while you have one for outbound calls. To get more detailed help an even an answer you have to provide more info. The cli output while trying to setup an internal call will help. Erik On 2 mrt 2010, at 00:31, carem gyssell nieto wrote: I have a problem with my internal extensions, I'm using Asterisk 1.6.2.5 and freePBX 2.6. When I call betwen extensions these don't connect. There is a long silence and finally hang up. I have an E1 whit r2 and I use openr2, but I don't have problems to do calls to the PSTN..my problem it only with internal extension. Please help...it's an Asterisk bug??. Thanks :S -- Carem Gyssell Nieto Garcia -- _ -- Bandwidth and Colocation 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] User on PC?
On Mon, 2010-03-01 at 23:46 +0100, Leif Neland wrote: I'm looking for a way for linux to query a pc if user X is on, and has used the pc recently or the screensaver is not active. If so, I'll route a call for user X to the phone near that PC. Ideas, anyone? 'who' can give you info who is logged in and when for all terminals on a linux machine. Also 'fgconsole' will be usefull. This assumes you got remote access (ssh probably) to the machine and you are able to execute commands as root (for the fgconsole at least) Check also the XDMCP protocol for the X Display Manager XDM, KDM, GDM etc (not sure which one your machines will be running) as it can provide some info also -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with install
hi, newbie. i installed from asteriskNOW cd onto a 2nd drive. using a dell system. has a 'c' drive. on reboot. it comes up 'GRUB and sits there. any clues? thanks. g. _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469226/direct/01/-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with install
ignore, i got it. the drive wasn't added in the bios. From: giv...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 1 Mar 2010 23:57:09 + Subject: Re: [asterisk-users] help with install hi, newbie. i installed from asteriskNOW cd onto a 2nd drive. using a dell system. has a 'c' drive. on reboot. it comes up 'GRUB and sits there. any clues? thanks. g. Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. _ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469227/direct/01/-- _ -- Bandwidth and Colocation 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 answer() necessary ?
Håkon Nessjøen wrote:
You only need to answer() the call when you want to play audio, or
music on hold, receive dtmf, etc.
If you are just sending the incoming call to a Dial() or Queue(without
music on hold), you don't need to answer.
The receiving party will do the answering. This way the callee doesn't
need to pay for the 'ringing'.
If the call is coming in on a DAHDI connection, and you have a special
agreement with your telco, you can do something called early audio.
To do this, you need to do Progress() call, and then remember to tell
the asterisk functions you are using, that they should not
automatically answer the call.
(Playback etc does this unless you explicitly tell it not to)
If you are connected to your telco via SIP, i'm not sure how this
works. I don't have any experience there. But I think it's about the
same.
About IVR-prompt and a queue.. Very few telcos, if any, let you
receive DTMF in early audio. I'm not sure if it is supported in
asterisk either. So I would say you always need to answer you call if
you are going to have IVR-prompts before a queue. Asterisk does this
for you anyways unless you tell it not to, when using functions like
Background and Playback.
Regards,
Håkon
On Mon, Mar 1, 2010 at 11:22 AM, jonas kellens jonas.kell...@telenet.be
wrote:
Hello list,
is it necessary to properly answer() an incoming call ?
I don't want to answer a call because the caller has to pay even if the
attached SIP-phones do not answer the phone call. Because I answer() the
incoming call, the caller has to pay for 60 seconds of 'ringtone'.
On the other hand, sometimes an incoming call is send to a macro where the
caller is given the opportunity to leave a voicemail message. It's to late
to answer() the call in the macro, but I guess the voicemail()-application
automatically anwers the call ??
How about an IVR-prompt and a queue ? Do I need to answer the incoming call
before playing a voiceprompt and before sending it into a queue ??
Greetingz,
Jonas.
--
Do you have to Answer() to reach the fax extension?
That is assume you have:
[incoming-pstn-line]
exten = fax,1,NoOp(Fax Detected) ;; the fax line
exten = fax,2,GoTo(incoming-fax,s,1)
exten = fax,n,Hangup();; the fax machine
exten =s,1,Answer() ; only answer after __ seconds
exten =s,n,Wait(3) ; wait to see if it's a fax
exten =s,n,Dial(${House_Phones},36) ; this should be six rings
Can I do away with the answer()? And the Wait() I assume?
sean
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Re: [asterisk-users] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid
On 26 February 2010 12:38, Trevor Peirce tpei...@digitalcon.ca wrote: Charles Wang wrote: The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com http://mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default insecure=port,invite Add sendrpid=yes here. The sip.conf of MYPBX likes below: [MYE1] type=peer host=mye1.abc.com http://mye1.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=did insecure=port,invite Add trustrpid=yes here. A. Why can't I receive the CALLERID from MYPBX(the secondary server)? I am sure I use Set(CALLERID(num) for it. B. Why does the CALLERID that sends from MYE1 become as Anonymous? How can I fix it with the correct orginal callerid(912345678)? C. Why does my FROM message become as Anonymous sip:anonym...@anonymous.invalid instead of 912345...@mye1.abc.com mailto:912345...@mye1.abc.com ? You see this because, even though the number has been made available to you, it's marked as a blocked call. Your server is honoring this and blocking the number when it dials the next server. By using Remote Party ID, you'll be able to carry this information forward to your next server. Or you could override the presentation using CallerPres function (in 1.6) or SetCallerPres dialplan command in previous versions... e.g. Set(CallerPres()=allowed) before doing the Dial to MYPBX. d -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help!!! Internal extensions not connect
On Tue, Mar 2, 2010 at 1:51 AM, lesouvage i...@meetmecall.nl wrote: You doesn't seem to have a proper context,extension,priority available for internal calls while you have one for outbound calls. To get more detailed help an even an answer you have to provide more info. The cli output while trying to setup an internal call will help. Erik Or maybe you have an internal dialplan built into the phone which block your call to the local extensions. As Erik suggest you have to provide more details HTH, Ioan -- _ -- Bandwidth and Colocation 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 answer() necessary ?
On Tue, Mar 2, 2010 at 4:17 AM, sean darcy seandar...@gmail.com wrote:
Do you have to Answer() to reach the fax extension?
That is assume you have:
[incoming-pstn-line]
exten = fax,1,NoOp(Fax Detected) ;; the fax line
exten = fax,2,GoTo(incoming-fax,s,1)
exten = fax,n,Hangup() ;; the fax machine
exten =s,1,Answer() ; only answer after __ seconds
exten =s,n,Wait(3) ; wait to see if it's a fax
exten =s,n,Dial(${House_Phones},36) ; this should be six rings
Can I do away with the answer()? And the Wait() I assume?
sean
Hello Sean,
In order to detect a fax session you have to answer first. A fax
sessions is detected based on the incoming fax tones received from the
calling party, when it tries to negotiate with the called party. Based
on the pattern of those tones you have also to Wait several seconds.
Conclusion: in case you need automatic fax handling you have to Answer + Wait.
Note: In order to have a good caller-id detection I recommend to have
also a Wait before the Answer - but maybe you do not have such
problems...
HTH,
Ioan.
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[asterisk-users] Does Asterisk 1.6.2.1 Support SIP TLS encryption
hi, all i want to realize more secure communication between asterisk sip end users. so i want to know Does Asterisk 1.6.2.1 Support SIP TLS encryption? if you can tell me same specific example to do encrypt, it's very appreciated. Thanks! -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Trunk with multiple remote ip-addresses
Hi, Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No problem to get outgoing calls to work but i have some problems with incoming. Did set srvlookup=yes in sip.conf. Sending all outgoing calls to sip-corporate.tele2.se which is either sip-corporate1.tele2.se (130.244.190.42) or sip-corporate1.tele2.se (130.244.190.46). If i do a sip show peer Tele2, I see that Asterisk has chosen one of them: ToHost : sip-corporate.tele2.se Addr-IP : 130.244.190.46 Port 5060 Now my problems starts, when Tele2 sends a call to my Asterisk, the call can come frome any of those two ip-adresses. If it comes from 130.244.190.46 everything if fine, but if it comes from 130.244.190.42: [Mar 2 08:46:03] NOTICE[1372]: chan_sip.c:19167 handle_request_invite: Failed to authenticate! I thought srvlookup=yes should take care about that, but then i read a little bit more and found: Note: Asterisk only uses the first host in SRV records. :( Can anyone plz give me some hint howto solve my problem? Regards, Magnus -- _ -- Bandwidth and Colocation 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
