[asterisk-users] PJSIP Real-time Text (T.140)
Hi, is the support of real-time text limited to the SIP channel driver only? Somehow Asterisk is not offering T.140 to the called party when initiating a call and including real-time text. In my pjsip.conf I allowed T.140 and enabled text support. Regards, Simon Hohberg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Empty user string on pjsip inbound trunk
Hi, I try to setup an inbound trunk using pjsip_wizard.conf. Now, when I receive a call from that trunk with an empty user string, and I try to match it with 's' in the dial plan, Asterisk reports that the extension was not found in the context. * pjsip_wizard.conf: [example] type = wizard sends_auth = no sends_registrations = no remote_hosts = example.com:5060 endpoint/context = from-extern * extensions.conf: [from-extern] exten => s,1,Playback(demo-thanks) * Asterisk log: res_pjsip_session.c: Call from 'example' (UDP:111.222.3.4:5060) to extension '' rejected because extension not found in context 'from-extern'. What am I doing wrong? Regards, Simon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP - Video Support for WebRTC
places, timer
Contact: <sip:6000@192.168.2.106:5060;transport=WS>
Content-Length: 0
<>
<--- SIP read from WS:192.168.2.103:49848 ---> SIP/2.0 488 Not
Acceptable Here
Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
To: <sip:bbglnljp@72rvpk435t95.invalid;transport=ws>;tag=tlkjkk4vl8
From: "6001" <sip:6001@192.168.2.106>;tag=as0ba3cd59
Call-ID: 5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060
CSeq: 102 INVITE
Supported: ice,replaces,outbound
Content-Length: 0
<->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.2.103:49848:
ACK sip:bbglnljp@72rvpk435t95.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
Max-Forwards: 70
From: "6001" <sip:6001@192.168.2.106>;tag=as0ba3cd59
To: <sip:bbglnljp@72rvpk435t95.invalid;transport=ws>;tag=tlkjkk4vl8
Contact: <sip:6001@192.168.2.106:5060;transport=WS>
Call-ID: 5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.8.2
Content-Length: 0
---
Scheduling destruction of SIP dialog
'5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060' in 32000 ms
(Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [6000@outgoing:2] Answer("SIP/6001-", "") in
new stack Audio is at 19538 Adding codec 13 (ulaw) to SDP Adding
codec 14 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding
codec 100030 (opus) to SDP
<--- Reliably Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 200
OK
Via: SIP/2.0/WS
0iemcrsq9tm0.invalid;branch=z9hG4bK7826783;received=192.168.2.103;rport=
49851
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
To: <sip:6000@192.168.2.106>;tag=as1792125e
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6408 INVITE
Server: Asterisk PBX 12.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:6000@192.168.2.106:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 1055
v=0
o=root 1700582523 1700582523 IN IP4 192.168.2.106 s=Asterisk PBX 12.8.2
c=IN IP4 192.168.2.106 t=0 0 m=audio 19538 RTP/SAVPF 0 8 9 109
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:109 opus/48000/2
a=fmtp:109
maxplaybackrate=48000;sprop-
maxcapturerate=48000;minptime=10;maxaveragebitrate=2;stereo=0;sprop-
stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=ptime:20
a=maxptime:60
a=ice-ufrag:7fbfa28012692271620bb8c22da32ff3
a=ice-pwd:30067a57115528082e8744df31454da4
a=candidate:Hc0a8026a 1 UDP 2130706431 192.168.2.106 19538 typ host
a=candidate:S57a9bd66 1 UDP 1694498815 87.169.189.102 19538 typ srflx
raddr 192.168.2.106 rport 19538 a=candidate:Hc0a8026a 2 UDP 2130706430
192.168.2.106 19539 typ host
a=candidate:S57a9bd66 2 UDP 1694498814 87.169.189.102 19539 typ srflx
raddr 192.168.2.106 rport 19539 a=connection:new a=setup:active
a=fingerprint:SHA-256
C1:E5:39:6E:FB:7A:97:5D:70:CF:65:EF:E7:5C:4D:37:1E:AE:9B:72:70:E4:C1:F4:
86:F8:7B:1A:8D:DE:B3:47
a=sendrecv
m=video 0 UDP/TLS/RTP/SAVPF 120 126 97
<>
<--- SIP read from WS:192.168.2.103:49851 ---> ACK
sip:6000@192.168.2.106:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK690056
Max-Forwards: 69
To: <sip:6000@192.168.2.106>;tag=as1792125e
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6408 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 2.0.2
Content-Length: 0
<->
--- (11 headers 0 lines) ---
<--- SIP read from WS:192.168.2.103:49851 ---> BYE
sip:6000@192.168.2.106:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK1426296
Max-Forwards: 69
To: <sip:6000@192.168.2.106>;tag=as1792125e
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6409 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 2.0.2
Content-Length: 0
<->
--- (12 headers 0 lines) ---
Scheduling destruction of SIP dialog '1ansppdrpdulbtr3j5ub' in 32000 ms
(Method: BYE)
<--- Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 200 OK
Via: SIP/2.0/WS
0iemcrsq9tm0.invalid;branch=z9hG4bK1426296;received=192.168.2.103;rport=
49851
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
To: <sip:6000@192.168.2.106>;tag=as1792125e
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6409 BYE
Server: Asterisk PBX 12.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
May you please help me to make it word? i'm just interessting for the
audio.
Thank you in advance
Hi Olivier,
I
Re: [asterisk-users] PJSIP Multipart Body
On 06/27/2016 12:09 PM, Joshua Colp wrote: Simon Hohberg wrote: Hi, I want to pass a part of a SIP INVITE multipart body. I found a quite old patch here: https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22 But this patch is for the SIP channel driver not PJSIP, right? Is it even possible without a patch? What do I have to put in the dialplan then? If you are asking if you can manipulate or get this information from the dialplan in PJSIP it's not currently possible. Hi Joshua, thank you for taking time to come back to me. It would be enough to just pass this body part on to the callee. What about the SIP channel driver, is there a way to do this? Regards, Simon -- _ -- 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] PJSIP Multipart Body
Hi, I want to pass a part of a SIP INVITE multipart body. I found a quite old patch here: https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22 But this patch is for the SIP channel driver not PJSIP, right? Is it even possible without a patch? What do I have to put in the dialplan then? Thanks in advance, Simon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?
Is it implied here that both HTTPS and WSS must also come from the same server (Same Origin Policy) ? No, the same origin policy does not apply to web sockets. Then, can I also install my own WebRTC demo page on my own private Asterisk server and access this demo page through HTTPS ? If I'm not mistaken, this should fulfill all requirements. It doesn't matter where the asterisk server is hosted. It is important where the web application comes from. If you don't want to use https and wss you only have the option to host the web app locally (on the same machine as the browser that loads the page), which probably makes sense only for development. Otherwise you have to use https and wss for the reasons discussed earlier. Hope it helps. Simon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?
Hi Oliver, On 02/18/2016 12:10 PM, Olivier wrote: Hello, I'm trying to have my first calls with WebRTC. My server has asterisk 13.7.0. I'm following the instructions from the wiki [1]. So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station. Whenever I type something like ws://123.123.123.123:8088/ws <http://123.123.123.123:8088/ws> in Expert Mode form (see [1]), I'm getting this error : *2:SecurityError: Failed to construct 'WebSocket': An insecure WebSocket connection may not be initiated from a page loaded over HTTPS.* If I replace ws://123.123.123.123:8088/ws <http://123.123.123.123:8088/ws> with wss://123.123.123.123:8088/ws <http://123.123.123.123:8088/ws>, this error message becomes with /Disconnected:*Failed to connet to the server*/ My questions are: 1. Is wss now required by sipml5 live demo (implying wiki page is not up-to-date) ? Yes, like the error says, you have to use wss on pages served via https. Furthermore, Chrome requires the use of https when you want to use getUserMedia. See here: https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en. It says: " Starting with Chrome 47, getUserMedia() requests are only allowed from secure origins: HTTPS or localhost." The solution for development is, to host the webrtc client locally, so that you load the page from localhost. In that case getUserMedia is allowed with http, too (as the quote says). That means you have to download the dubango client and run a webserver on your dev machine. 2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ? Unfortunately, there is not much documentation about this, as far as I can tell. Regards [1] https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 [2] https://www.doubango.org/sipml5/ Regards, Simon -- _ -- 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] Delayed start of video with WebRTC - Missed FIR due to DTLS?
Hi, I am using Asterisk 13.7.0 with PJSIP. I set up Asterisk for use with WebRTC SIP clients. After I managed to get video working, I noticed, that the calling party receives no video till 90s (or so) have passed. After 90s both parties receive video perfectly. I am suspecting that this is due to the time needed for the DTLS handshake between Asterisk and the caller. Since Asterisk first establishes a full connection to the callee, the callee already begins sending data, while Asterisk is still performing the DTLS handshake with the caller. As a consequence the caller misses the first RTCP Full Intraframe Request (FIR) and the received video stream cannot be rendered till the next FIR 90s later arrives. Am I right or is this nonsense? Is this a known issue? I couldn't find anything about this. Is there a fix available? Thanks in advance! Simon -- _ -- 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] load-balancing AMI and load-balancing FastAGI?
Anyone? -- _ -- 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] load-balancing AMI and load-balancing FastAGI?
Hi, I am starting a new project to develop a predictive dialler system. - Agents can start receiving calls from the queue if agent press Available button on the browser which will unpause the queue on Asterisk. - About 100-150 concurrents calls on a Asterisk box - Call-out initiated. Other end answers. Passes AMD. Lands in Queue and direct to agents that is available and call is recorded. - Update state of the call (Ringing, Talking, etc) on the database. - Listen the events such as Hang Up from customer, check if call is successfully originated or what the failure, etc. - Agent will have ability to transfer customer call to other agent or external number. As described above to develop a predictive dialler system, is it best to use AMI or FastAGI? I am aware that I can setup FastAGI load balancing such as agitator (FastAGI reverse proxy). AMI case: load-balances incoming events/response across multiple processes (multiple AMI connections on the same asterisk machine), should the ami events/response should be pushed into RabbitMQ so the proess can read from RabbitMQ ? Thanks Paul -- _ -- 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] res_fax.c: allowed rates for V27 modems
Hi all, We are running a fax2email service based on asterisk 1.8.18.0, and we are currently trying out asterisk 1.8.32.2 in our labs. We get the following error when sending faxes out: [Apr 7 14:34:20] ERROR[16653]: res_fax.c:2121 sendfax_exec: 'modems' setting 'V17,V27,V29' is incompatible with 'minrate' setting 2400 It looks like function check_modem_rate in res_fax.c has been updated and rate 2400 is not allowed for V27 any more. Found the following issue explaining the change: https://issues.asterisk.org/jira/browse/ASTERISK-23231. However ITU-T specifications for V27ter (which should supersede the one for V27) specify both 2400 and 4800. We are currently receiving faxes at rate 2400 on our production servers so we can't upgrade asterisk as is. Does anybody have some insights on this? 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
[asterisk-users] Dialplan for receiving faxes on Asterisk
Hi all, It looks like people commonly use this kind of dialplan when receiving faxes on Asterisk, with a jump to extension fax during the Wait() if a fax tone is detected: [start-here] exten = _X.,1,Answer() exten = _X.,n,Wait(n) exten = _X.,n,...do stuff... exten = _X.,n,Hangup() exten = fax,1,Goto(fax-rx,receive,1) [fax-rx] exten = receive,1,... exten = receive,n,...do stuff... exten = receive,n,ReceiveFAX() This is well suited in case Asterisk needs to receive both voice and fax calls. But what if Asterisk is only used to receive fax calls, can we start directly at the fax-rx context? I've heard that it's better to wait a few seconds before calling ReceiveFAX(), is it still necessary in case we don't actually need fax detection? Simon -- _ -- 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] [DAHDI] qozap instead of wcb4xxp
Hello, I have a Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] card which seems to work on 1 machine but not on another. It SHOULD load this driver: dahdi_hardware -v pci::00:00.0 wcb4xxp+ 1397:08b4 Junghanns QuadBRI ISDN card Instead of: dahdi_hardware -v pci::00:00.0 qozap- 1397:08b4 Generic Cologne ISDN card There is no qozap modul in lib/modules but I tried to blacklist it anyway. The consequences that the card is not working in the second machine with the same dahdi config/kernel: Sarting background readahead: [ OK ] Checking for hardware changes [ OK ] Starting dahdi: Loading DAHDI hardware modules: wct4xxp: [ OK ] wcb4xxp: [ OK ] No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: DAHDI_SPANCONFIG failed on span 1: No such device or address (6) [FAILED] [FAILED] lspci = 0:00.0 Non-VGA unclassified device: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Control: I/O- Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Region 0: I/O ports at 4000 [disabled] [size=8] Region 1: Memory at e020 (32-bit, non-prefetchable) [disabled] [size=4K] lsmod = Module Size Used by bluetooth 53925 2 ipv6 267617 19 xfrm_nalgo 13381 1 ipv6 crypto_api 12609 1 xfrm_nalgo autofs429253 3 i2c_dev12613 0 i2c_core 23745 1 i2c_dev lockd 63209 0 sunrpc145533 2 lockd wcb4xxp76708 0 wct4xxp 348896 0 dahdi 196296 4 wcb4xxp,wct4xxp crc_ccitt 6337 1 dahdi loop 18889 0 dm_mirror 24521 0 dm_multipath 24909 0 scsi_dh11713 1 dm_multipath scsi_mod 142485 1 scsi_dh parport_pc 29157 0 lp 15849 0 parport37641 2 parport_pc,lp pcspkr 7105 0 xennet 29769 0 [permanent] dm_raid45 67273 0 dm_message 6977 1 dm_raid45 dm_region_hash 15809 1 dm_raid45 dm_log 14657 3 dm_mirror,dm_raid45,dm_region_hash dm_mod 63225 4 dm_mirror,dm_multipath,dm_raid45,dm_log dm_mem_cache9921 1 dm_raid45 xenblk 20149 3 ext3 124873 2 jbd57065 1 ext3 uhci_hcd 25677 0 ohci_hcd 24937 0 ehci_hcd 34253 0 /etc/init.d/dahdi stop Unloading DAHDI hardware modules: ERROR: Module dahdi is in use error + /etc/init.d/dahdi start Loading DAHDI hardware modules: wct4xxp: [ OK ] wcb4xxp: [ OK ] No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: DAHDI_SPANCONFIG failed on span 1: No such device or address (6) [FAILED] /usr/sbin/dahdi_cfg - DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 12) 12 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) Putting the card to other slot doesn't help. That would be nice if some of the developers of this hardware would be around here. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
[asterisk-users] External call control for Asterisk
Hi there, I’m new to Asterisk and there’s a ton of documentation. I’m not really sure where to start. What I want to do is this: a PBX service ala FreePBX, but where call control is passed via SIP to an external service which will tell Asterisk: a) * Whether the call is allowed b) * Where to connect the call, if necessary (i.e. forced redirection to a C-party) c) * To disconnect the call at some time in future based on charging considerations (i.e. online charging) There is also the option of not using Asterisk at all, and simply using the other service directly, but Asterisk is much better suited to handling end-user devices. The external service does control logic only. Can someone point me at the right place in the documentation to get a handle on where I should be hooking things like this? -- Cheers Simon -- _ -- 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] External call control for Asterisk
On Wed, 10 Apr 2013, Simon Green wrote: Hi there, I’m new to Asterisk and there’s a ton of documentation. I’m not really sure where to start. What I want to do is this: a PBX service ala FreePBX, but where call control is passed via SIP to an external service which will tell Asterisk: a) * Whether the call is allowed b) * Where to connect the call, if necessary (i.e. forced redirection to a C-party) c) * To disconnect the call at some time in future based on charging considerations (i.e. online charging) It depends... You could probably do all of this just using dialplan logic and Asterisk's internal database. If you are looking to build a system that will scale, you'll want to store your call processing parameters in a database like MySQL and access the parameters using an AGI (an external program written in any language you are comfortable with) and then write a dialplan to follow your business logic. While the dialplan language does include methods to access databases, I find it cumbersome, limited, and ugly. I like to keep all the nasty details in a little black box and keep my dialplan clean and maintainable. It actually looks a little like I might be better off front-ending with OpenSIPS, which can do AAA via Diameter, and then passing the call, once allowed, to Asterisk. I'm certainly keen to put extension provisioning information into MySQL, but I need the realtime accounting aspect as much as the authorisation aspect. Ideally I'm after Asterisk to be, effectively, a smart media gateway. I want it to handle basic registration of user clients, but for realtime call authorisation and sometimes routing to be handled off-board. Effectively, a prepaid calling service with Asterisk handling the calling and the other system handling the prepaid. Am I barking up the wrong tree? -- _ -- 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] Need help defining a stackexchange (i.e. stackoverflow) for telephony
It's nearly there now, just need a few more votes in order for it to trigger the next phase. Please take a moment to vote if you're interested: http://area51.stackexchange.com/proposals/12932/telephony/ On Mon, 9 May 2011, Simon P. Ditner wrote: For those of that are fans of stackoverflow.com, and stackexchange.com, there's an effort to define a telephony stackexchange site. It's still in the definition phase. What it needs to move forwards is more votes on on/off topic questions, and perhaps some better questions to vote for or against. If you're interested in helping out, or following the progress, visit: http://area51.stackexchange.com/proposals/12932/telephony/ Cheers, spd -- _ -- 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] Need help defining a stackexchange (i.e. stackoverflow) for telephony
For those of that are fans of stackoverflow.com, and stackexchange.com, there's an effort to define a telephony stackexchange site. It's still in the definition phase. What it needs to move forwards is more votes on on/off topic questions, and perhaps some better questions to vote for or against. If you're interested in helping out, or following the progress, visit: http://area51.stackexchange.com/proposals/12932/telephony/ Cheers, spd -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SS7 error
Hi Asterisks Team, I am getting the error below after getting a connection to a telco using ss7. Anyone know how to solve it? The link keeps coming up and down every 30 seconds. Resetting CIC 3 [Mar 28 15:16:28] WARNING[20434]: chan_dahdi.c:11913 ss7_linkset: RSC on unconfigured CIC 3 Received out of sequence MSU w/ fsn of 119, lastfsnacked = 116, requesting retransmission MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. Resetting CIC 4 [Mar 28 15:16:28] WARNING[20434]: chan_dahdi.c:11913 ss7_linkset: RSC on unconfigured CIC 4 Received out of sequence MSU w/ fsn of 119, lastfsnacked = 117, requesting retransmission MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. Thanks for your help in advance. FYI am using Asterisks 1.6 Peter. -- _ -- 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] Asterisks with ss7 problem
Hi, I am trying to set up asterisk with ss7. Whenever I try to load module chan_dahdi.so, I get the error [Mar 26 17:33:27] ERROR[10437]: chan_dahdi.c:10458 mkintf: Unable to find linkset -1 I have compiled dahdi, libss7, asterisks (am using asterisk 1.6) in that order. Have already set signalling to ss7 in dahdi_channels.conf How do I sort this out? Thanks for your help in advance. Peter. -- _ -- 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] phone emulator for doing interop testing
I'm looking for some software to do emulation of phones so that I can test out a whole range of phones and their firmware revisions against asterisk. Anyone know of something like that? I'm hoping that the hardware vendors have something like that, and _maybe_ I'm really lucky and they are using something somewhat standard like qemu. Cheers, spd -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner ___ -- 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] DTMF tones mid conversation
-u397 P[ 1] -- * Unknown Indication:20 pid:168 P[ 1] -- * Unknown Indication:20 pid:168 P[ 1] I IND :DISCONNECT oad:07x dad:78xxx pid:167 state:CONNECTED P[ 1] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:78xxx P[ 1] -- info_dad: onumplan:2 dnumplan:0 rnumplan: cpnnumplan:0 P[ 1] -- queue_hangup P[ 1] I SEND:RELEASE oad:07x dad:78xxx pid:167 P[ 1] -- channel:1 mode:TE cause:16 ocause:-1 rad: cad:781890 P[ 1] -- info_dad: onumplan:2 dnumplan:0 rnumplan: cpnnumplan:0 ...Unknown indication 20 of any relevance? Is anyone able to confirm exactly whether mISDN's hardware DSP and driver is responsible for detecting DTMF, or whether it's Asterisk analysing the inbound audio? Scanning the README.misdn (sourced separately) the chan_misdn driver readme comments a feature as DTMF Detection in HW+mISDNdsp (much better than asterisks internal!) - so surely DTMF is recognised and passed on by mISDN to Asterisk. The fact that the log messages prefixed by P[ 1] are mISDN - I think I've answered my own question there... Prior to going down the mISDN route, I looked at Dahdi as the Dahdi configs mention native Dahdi B410P support. But, the conclusion I came to (although what I read didn't make it clear me) is that the readme was referring to Dahdi B410P support in Ast 1.6, not 1.4. That sound right? Dahdi readme: Installing the B410P drivers with mISDN ~~~ DAHDI includes the wcb4xxp driver for the B410P, however, support for the B410P was historically provided by mISDN. If you would like to use the mISDN driver with the B410P, please comment out the wcb4xxp line in /etc/dahdi/modules. This will prevent DAHDI from loading wcb4xxp which will conflict with the mISDN driver. Enough reading.. if you're still awake! Any help would be very much appreciated. Thank you, Simon Date: Wed, 11 Feb 2009 14:58:58 + From: paulo.r.san...@sapo.pt To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF tones mid conversation Andrew Thomas wrote: I seem to have a problem of intermittent DTMF tones being played during a conversation. I'm having the same problem, but in my case, it's every 1 minute and at the start of the call. I wonder if it has anything to do with echo cancellation. I've only noticed when using a Zap channel, but I'll run some more tests. asterisk 1.4.17 / addons 1.4.7 / zaptel 1.4.12.1 / mISDN 1.1.8 Paulo Santos ___ -- 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 _ Check out the new and improved services from Windows Live. Learn more! http://clk.atdmt.com/UKM/go/132630768/direct/01/___ -- 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] Siemens S685IP registration problems
Hi folks, I wonder if any of you out there are using Siemens S685IP base station(s) (with S68H handsets) on Asterisk and experiencing problems with SIP registrations where the SIP extensions do not ring and peers become unreachable after a period of time. Symptoms are rather sporadic, but as described, SIP extensions being unreachable from Asterisk perspective. Also experienced 'not possible' messages trying to dial using the handsets. In Asterisk - 'sip show peers' shows the SIP device status as 'UNKNOWN' and console output says chan_sip.c Peer 'xxx' is unreachable. (as one would expect if it can't see it!!) So I'm thinking it's a problem with the base.. or.. some issue with qualifications and possibly the base station not responding (guessing here). I'm finding that the Siemens web GUI reports messages similar to 'server not accessible' or 'registration failed'. These messages appear randomly throughout the day following successful previous registration. Have enabled 'sip set debug ip xxx' for one of the bases and on the Siemens web GUI disabling the SIP account and re-enabling it doesn't work generally, and no SIP messages are being directed from the base to Asterisk. Leads me to think it's a base/firmware issue. Some times the phones are contactable for a day without fault, other times they're problematic at random intervals. It's not always all of the SIP accounts assigned on the Siemens base, sometimes it's just one account, other times it's all accounts. (What a horrible situation to debug/fault find! Glad my Aastra's are reliable!) The only resolve I've found which is rather unacceptable is to reboot the Siemens base station. Upon doing so, the base re-registers all of the accounts to the Asterisk server and calls to/from handsets work for 'a period of time'... My setup is as follows: - Asterisk 1.4.22 - Base 1 has 3x handsets and 3x SIP accounts (providers) and the SIP accounts are individually assigned to each handset. - Base 2 has 5 handsets and 6 SIP accounts. 5 SIP accounts for each handset, then the 6th SIP account is a 'group' extension which rings all of the phones on the base station. MWI for VM is also used and works. Call waiting disabled on handsets. - Both bases on latest available as of Jan 09 - 0214 / 043.00 The bases are set up with static IPs, info services off, etc. Am not using SIP domains, no NAT, all communicating on a LAN on same network so no routing or latency issues. Registrar server defined and refresh time set to 180 seconds (Siemens default). SIP.conf has nat=no, qualify=yes. host= is currently dynamic.. maybe I should set this to the IP of the base as they're using static IPs, but reading the specs of this setting describe set to 'dynamic' if the phone should register itself... hmm. I’ve seen similar posts from other users on the exact same subject. Sources as follows: Voip-Info http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP (see 'Discussions' tab). The Open Sourcerer: http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/comment-page-3/ Siemens Forums: http://www.siemens-gigaset-forum.com/de/posts/list/13788.page Siemens Customer care: http://www.siemens-gigaset-forum.com/de/posts/list/13788.page (people with aged open support calls) Siemens support via phone were rather unhelpful and didn't grasp the technicalities of the issue I was conveying so drew a blank (have I checked my router.. hmm!) I'm guessing this is a firmware issue but intrigued to know if others are experiencing the same. Anyone else experiencing similar problems? Or indeed successes with a similar setup? Can anyone recommend a stable working DECT SIP phone for enterprise use with Asterisk? (The Snom M3 looks good but read about issues with transfers which concern me) May have to resort to ditching the S685s and go with Aastra desk sets all round - shame to lose the flexibility of cordless though. Many thanks in advance. _ Cut through the jargon: find a PC for your needs. http://clk.atdmt.com/UKM/go/130777504/direct/01/___ -- 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] Calls drop after a couple of minutes.
I have been encountering a rather hard to debug problem for the last couple of months: * Calls are setup fine. * After a couple of minutes, two way audio becomes one-way and the remote or local party drops out of the call. Setup: * Nokia E71i sip on NAT'd network (multihomed linux box) * Remote asterisk 1.4.21 on Ubuntu on public network * using a Finera/Betamax provider to route calls to PSTN. I initially thought it may be a NAT problem and have checked everything on the NAT gateway/firewall. I see no rejected packets hitting the firewall logs. I'm really at a loss as to what could be causing the calls to drop out for one party so regularly. Any clues where I could look further to debug this would be most useful. local firewall: modprobe ip_conntrack_sip ports=5060 modprobe ip_nat_sip # probably not needed since everything is forwarded: $IPTABLES -A FORWARD -s $INTERNAL_NET -d $ANYWHERE -p udp --dport 5060 -j accept-log # sip remote Asterisk server: $MODPROBE ip_conntrack $MODPROBE ip_conntrack_sip ports=5060 $IPTABLES -A INPUT -s $ANYWHERE -d $PUBLIC_ADDR -p udp --dport 5060 -j accept-log # voip $IPTABLES -A OUTPUT -s $PUBLIC_ADDR -d $ANYWHERE -p udp --sport 5060 -j accept-log # voip $IPTABLES -A OUTPUT -s $PUBLIC_ADDR -d $ANYWHERE -p udp --dport 5060 -j accept-log # voip $IPTABLES -A INPUT -s $ANYWHERE -d $PUBLIC_ADDR -p udp --dport 1:2 -j accept-log # voip $IPTABLES -A OUTPUT -s $PUBLIC_ADDR -d $ANYWHERE -p udp --sport 1:2 -j accept-log # voip sip.conf: [101] callerid=Simon Tennant type=friend username=101 secret=xx host=dynamic reinvite=no canreinvite=no mailbox=101 context=from-internal nat=yes port=5060 qualify=yes insecure=very disallow=all allow=alaw also sip.conf [justvoip.com] type=peer host=sip.justvoip.com fromdomain=sip.justvoip.com progressinband=yes disallow=all allow=alaw ; only alaw works with sip1... nat=no canreinvite=no qualify=yes insecure=port,invite username=imagi-justvoip fromuser=00491785450880 secret= registerattempts=0 ; keep trying to register (normally times out after 10 attempts) context=from-external from rtp.conf rtpstart=19000 rtpend=2 -- Simon Tennant _ fixed: .uk +44 20 7043 6756 .de +49 89 420 955 854 mob: .uk +44 78 5335 6047 .de +49 17 8545 0880 xmpp: [EMAIL PROTECTED] ___ -- 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] Elastix workshop in Toronto; Wed Nov 26th, 2008
This Wednesday, November 26th, the Toronto Asterisk Users Group invites all in the area to join us for a telephony workshop and talk sponsored by Sangoma Inc.[1] Jose Landivar, co-founder of PaloSanto Solutions[2], creators of Elastix, will be running a getting started workshop on Elastix, followed by a talk discussing how it differs from other Asterisk-based distributions, and a road map of the project's future. Elastix[3] is an open source asterisk-based linux telephony appliance that integrates tools such as OpenFire IM Server, SugarCRM, mail services, and billing software into a single, easy-to-use interface. It also adds its own set of utilities and allows for the creation of third party modules. When: Wednesday November 26th, 2008 5:00 pm - 7:00 pm: WORKSHOP - Getting Started with Elastix (reg. req.) 7:00 pm - 8:00 pm: TALK - Integrated Communications with Elastix Where: Committee Room 3 North York Civic Centre (in Mel Lastman Square) 5100 Yonge St., North York, ON Map link: http://xrl.us/hqbw Registration is requested for the workshop, sign up at: http://taug.ca/node/174 No registration is required for the talk: http://taug.ca/node/175 Check back at http://taug.ca for event updates. Cheers, Simon P. Ditner TAUG.ca Talk Coordinator [1] http://www.sangoma.com [2] http://www.palosanto.com [3] http://www.elastix.org ___ -- 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] Slighly OT?.. headset for Linksys SPA922
Hi there, Is anyone using a headset with one of these phones? If so, can you recommend any? Thanks Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922
So any 2.5 headset will work with the SPA922? On Fri, Aug 1, 2008 at 12:23 PM, Paul Hales [EMAIL PROTECTED] wrote: Plantronics. PaulH Simon wrote: Hi there, Is anyone using a headset with one of these phones? If so, can you recommend any? Thanks Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stop vm-intro being played
Hi There, Is there a way just to have the custom voice message play, and not have asterisk play: vm-intro after that? Thanks Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stop vm-intro being played
Got it!... we are using Elastix and i just had to set s in the VM_OPTS in the extensions_additional.conf file. On Sun, Jul 20, 2008 at 1:41 PM, Simon [EMAIL PROTECTED] wrote: Hi There, Is there a way just to have the custom voice message play, and not have asterisk play: vm-intro after that? Thanks Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AVM Fritz BRI cards and echo cancellation
Hi There, We are using 2 x AVM Fritz BRI cards with mISDN. The phones are Linksystem SPA922's and we are getting a little echo on the lines.. from what i unserstand, these are passive cards and do not have any onboard echo cancellation, but im wondering if there is anything that can be done software/config wise to help with this? I did find this (http://www.misdn.org/index.php/FAQ): 4) You set another value for tx-gain, -1 for example to prevent echoes. Please set the tx-gain back to 0 for those calls as in 3) (vt0). Can anyone comment on this please? Thanks Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 AVM ISDN Fritzcards
On Thu, Jul 3, 2008 at 5:07 PM, Dave Cotton [EMAIL PROTECTED] wrote: Yes, with Suse 10.2/10.3 and chan_misdn. Just to follow up on this. SLES 10.2 SP2 worked bang on. The two cards are configured and working correctly and recognised by Asterisk. Question: I guess you were meaning openSUSE 10.2/10.3... will openSUSE 11 work here? Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 AVM ISDN Fritzcards
On Thu, Jul 3, 2008 at 5:07 PM, Dave Cotton [EMAIL PROTECTED] wrote: Simon wrote: Hi There, Has anyone managed to get 2 AVM ISDN Fritzcard's working in with a 2.6 kernel system? Yes, with Suse 10.2/10.3 and chan_misdn. OK. ive got debian etch working with one card compiling the drivers from AVM, but beyond that im having trouble getting the next card working - its a bit beyond my level here. Im guessing that Suse 10.3 is 'just going to work' in this case? rather than having to compile drivers etc? Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 AVM ISDN Fritzcards
Hi There, Has anyone managed to get 2 AVM ISDN Fritzcard's working in with a 2.6 kernel system? Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Config help with ISDN Fritzcard
Hi There,
Ive managed to get a AVM ISDN Fritzcard working with debian etch (see
capiinfo output below), and compiled chan_capi and got everything
finished (i think). So i have: Etch + Asterisk + Zaptel + ChanCapi +
Asterisk Addons + Asterisk-GUI and the chan_capi driver is loaded into
asterisk:
asterisk*CLI module show like capi
Module Description
Use Count
chan_capi.so Common ISDN API Driver (1.1.1) 0
1 modules loaded
Would someone be able to help me with the config to setup the incoming
calls from the ISDN card? I dont know where to start here.
Thanks
Simon
asterisk:~# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-07 (49.23)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
Modem
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
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[asterisk-users] Disto choice for Asterisk with AVM Fritz!PCI cards
Hi There, I am looking to build an Asterisk server with dual AVM Fritz!PCI cards linked to 2 BRI in New Zealand. Just wondering if anyone has done this, and if you have any ideas about the best disto choice for this task? Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] measuring network quality in the field
What open source tools are people using to quantitatively measure how well QoS/traffic shaping is performing out in the field, and what call quality people are experiencing in terms of jitter and packet loss? Cheers, spd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X-Lite and Presence?
Cool - thanks Rob. I will check it out tmorrow. Simon On Wed, Apr 16, 2008 at 4:34 PM, Rob Hillis [EMAIL PROTECTED] wrote: IIRC Asterisk doesn't support the full presence publishing spec so you won't get the full range of possible status types, however you should at least get free/busy. I vaguely recall having to change the presence type from peer-to-peer to something else - that's done in the SIP configuration window. However, since I don't have X-Lite in front of me at the moment (fortunately, for the most part!) I can't give you more of a hint than that. Simon wrote: Thanks again!.. Right. I have it working now, it shows the users statuses as online or offline and changes them when someone closes their app. But not free/busy type changes.. Any idea why here? Simon On Wed, Apr 16, 2008 at 3:21 PM, Rob Hillis [EMAIL PROTECTED] wrote: X-Lite. Of course, Asterisk will need a hint configured for that extension as well... Simon wrote: Thanks for the reply.. Sorry for the lame question.. Do i do that in X-Lite or Asterisk? On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis [EMAIL PROTECTED] wrote: Configure the extension as a softphone using the format extension@asterisk.ip.address. Works fine for me - and works even better for agents! Simon wrote: Hi There, We have some users using x-lite as their SIP phone... but im wondering how to get the Calls Contacts to show as being available (Or if it can be done at all?). Is this what Presence is? Thanks Simon ___ -- 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 ___ -- 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 ___ -- 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 !DSPAM:48057d5e261007514015341! ___ -- 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
[asterisk-users] QOS for outgoing SIP calls
Hi There, We have our Asterisk box using a external SIP provider for outgoing calls over our DSL line. This seems to be going well... But i do have the ability to set some QOS ports in our linksystem DSL router... Its faily basic, so im wondering if it will help at all... We can specify High, Med, Low settings for: FTP, HTTP, Telnet, SMTP and POP3. Plus we have the ability to specify up to 3 ports for the same settings. Is this worth doing? If so, what ports should i specifiy? Simon ___ -- 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] X-Lite and Presence?
Hi There, We have some users using x-lite as their SIP phone... but im wondering how to get the Calls Contacts to show as being available (Or if it can be done at all?). Is this what Presence is? Thanks Simon ___ -- 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] X-Lite and Presence?
Thanks for the reply.. Sorry for the lame question.. Do i do that in X-Lite or Asterisk? On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis [EMAIL PROTECTED] wrote: Configure the extension as a softphone using the format extension@asterisk.ip.address. Works fine for me - and works even better for agents! Simon wrote: Hi There, We have some users using x-lite as their SIP phone... but im wondering how to get the Calls Contacts to show as being available (Or if it can be done at all?). Is this what Presence is? Thanks Simon ___ -- 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 !DSPAM:48055196261001131342032! ___ -- 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] X-Lite and Presence?
Thanks again!.. Right. I have it working now, it shows the users statuses as online or offline and changes them when someone closes their app. But not free/busy type changes.. Any idea why here? Simon On Wed, Apr 16, 2008 at 3:21 PM, Rob Hillis [EMAIL PROTECTED] wrote: X-Lite. Of course, Asterisk will need a hint configured for that extension as well... Simon wrote: Thanks for the reply.. Sorry for the lame question.. Do i do that in X-Lite or Asterisk? On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis [EMAIL PROTECTED] wrote: Configure the extension as a softphone using the format extension@asterisk.ip.address. Works fine for me - and works even better for agents! Simon wrote: Hi There, We have some users using x-lite as their SIP phone... but im wondering how to get the Calls Contacts to show as being available (Or if it can be done at all?). Is this what Presence is? Thanks Simon ___ -- 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 ___ -- 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 !DSPAM:48056806261001804221289! ___ -- 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
[asterisk-users] CallerID in NZ
Hi There, We have a Asterisk 1.4 box with a X100P card connected to a analog line with Caller ID serrvices enabled on it. When an incoming call appears we get the following in the log: -- Starting simple switch on 'Zap/1-1' -- Detecting post-CID distinctive ring [Apr 15 10:38:07] NOTICE[7151]: chan_zap.c:6469 ss_thread: Got event 18 (Ring Begin)... [Apr 15 10:38:07] NOTICE[7151]: chan_zap.c:6469 ss_thread: Got event 2 (Ring/Answered)... [Apr 15 10:38:07] NOTICE[7151]: chan_zap.c:6469 ss_thread: Got event 18 (Ring Begin)... -- Detected ring pattern: 299,290,279 -- Checking 0,0,0 -- Checking 0,0,0 -- Checking 0,0,0 -- Executing [EMAIL PROTECTED]:1] ExecIf(Zap/1-1, 0|SetCallerPres|unavailable) in new stack -- Executing [EMAIL PROTECTED]:2] ExecIf(Zap/1-1, 0|Set|CALLERID(all)=unknown 000) in new stack So its not seeing the caller id. What might i have incorrect here? Thanks Simon ___ -- 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] The most efficient way to know SIP phones IP addresses ?
You could try:
asterisk -rx database get SIP/Registry 101 | cut -f 2 -d ':'
Which is not much shorter, but probably more efficient
Simon Elliston Ball
[EMAIL PROTECTED]
http://www.simonellistonball.com/
On 31 Mar 2008, at 10:02, Olivier wrote:
Hi,
Sometimes, you need to send requests to SIP phones either from Linux
command line or from Asterisk dialplan.
Which is the most efficient way to know a SIP phone IP address ?
Today, I think I would use :
asterisk -rx sip show peer 692 | grep Addr-IP | awk '{print $3}'
I'm wondering if anything more concise and efficient exists ?
Regards
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Re: [asterisk-users] The most efficient way to know SIP phones IP addresses ?
The asterisk database system is really more of a hash table than a
full database, so it's unlikely to happen. It's actually berkeley db
underneath.
Of course you could always create your own table on calls by using
something like Set(DB(ips/692)=${SIPPEER(692|ip)}) in the dialplan,
but it's probably a lot easier to just use the registry database, just
depends on how often you're going to be doing the lookups.
simon
Simon Elliston Ball
[EMAIL PROTECTED]
http://www.simonellistonball.com/
On 31 Mar 2008, at 10:56, Olivier wrote:
2008/3/31, Simon Elliston Ball [EMAIL PROTECTED]: You
could try:
asterisk -rx database get SIP/Registry 101 | cut -f 2 -d ':'
Which is not much shorter, but probably more efficient
That's fine !
Too bad one cannot input more specific database queries such as
database get SIP/Registry/Addr-IP 101.
Simon Elliston Ball
[EMAIL PROTECTED]
http://www.simonellistonball.com/
On 31 Mar 2008, at 10:02, Olivier wrote:
Hi,
Sometimes, you need to send requests to SIP phones either from Linux
command line or from Asterisk dialplan.
Which is the most efficient way to know a SIP phone IP address ?
Today, I think I would use :
asterisk -rx sip show peer 692 | grep Addr-IP | awk '{print
$3}'
I'm wondering if anything more concise and efficient exists ?
Regards
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[asterisk-users] Netgear TA612V line 2 and asterisk
Hi all, I have a Netgear TA612V voip adapter which I am trying to convince to work with asterisk. If I activate one of the two lines (line one or line two). The unit registers with the server no problem. If I try to register both lines with different usernames passwords the registrations fail and the server responds with 401 unauthorized. A search of the web shows that this has been seen before with other multi-line Netgear VOIP products. It is perhaps due to the fact that the source and destination ports for the SIP channel for both lines are always the same values (default 5060) meaning as far as the server is concerned the IP communication is going down the same logical connection between VOIP adapter and Asterisk server. Has anyone else seen this and is there either a work around or fix? Many thanks Simon -- Simon Falvey ___ -- 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] Astersik Transcoder support
http://www.digium.com/en/products/voice/tc400b.php Simon Elliston Ball [EMAIL PROTECTED] On 1 Feb 2008, at 17:29, Charles Feng wrote: Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles Never miss a thing. Make Yahoo your homepage. ___ -- 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] Free IAX / SIP Softphone with attended transfer
Zoiper is pretty impressive, it's a simple, neat little client. The one problem I have with it is the keyboard. I've had problems trying to use the keyboard to send DTMF on the current call. The left hand popout keypad is also a little small for my users' taste. It would be nice to have a keyboard hang-up, something like ESC, ditto for things like cancel buttons around the app. I really like the fact it does both SIP and IAX. Onto sillier issues: the icon is nice, but it would be great to have proper gamma anti-aliasing on the mac one. Just my .02 on the free mac os version, I might have to check out the biz edition too. It's all looking good. Good luck with the next release! Simon Simon Elliston Ball [EMAIL PROTECTED] On 23 Jan 2008, at 08:35, Zoa wrote: You can find it here: http://www.zoiper.com/downloads/free/linux/zoiper201-linux.tar.gz Note that the linux version does not support TLS and SRTP yet. * Instructions: * 1) Download zoiper201-linux.tar.gz 2) Extract Zoiper. If you don't use a GUI application for archive processing, here is the command line: tar zxf zoiper201-linux.tar.gz ./zoiper 3) Start Zoiper. *ZoIPer depends on ALSA library, so it* **must** *be installed! * Zoa Robert Moskowitz wrote: zoa wrote: Have you tried our Zoiper softphone yet (www.zoiper.com) - new version scheduled for in a couple of days ? If so, can you send me any remarks of list so that we can keep those things in mind for future versions ? Do you know where I can get it as an rpm to install on Centos 5 with Gnome? I do not have the time resources to do compiles. I am really a security protocol researcher and would be very interested in seeing what you have done for SIP TLS and SRTP. But for the later, I am all Linux. The one XP system is a corp box that I cannot add any software too. ___ -- 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] Asterisk desktop tools for OS X
Looks interesting. I couldn't get it working because a few of the preference fields were not responding (current svn, build on Leopard). Looks like a nice elegant solution though. Let me know if there's anything you want help on and I'll dust off my cocoa! Simon Simon Elliston Ball [EMAIL PROTECTED] On 17 Jan 2008, at 13:06, Lito Manansala wrote: Hi, Im interested, Please send me copy Thanks On Jan 17, 2008 7:25 PM, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have been long working on a project ( http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- 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 -- Regards, Lito Manansala Network Operations (VoIP) VoiceValley Group of Companies Phone: +61-7-30188461 Fax: +61-7-30188499 ___ -- 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] asterisk to mysql database!
Try: http://www.voip-info.org/wiki/view/Mysql and the links thereon. simon Simon Elliston Ball [EMAIL PROTECTED] On 16 Jan 2008, at 19:11, Naveen Palani wrote: Hello, Is there a possibility to connect from asterisk to mysql database without the interface application like Ruby or PHP. If i can connect to mysql database from asterisk, i can update the database for manipulations. Appreciate your response. Regards, Naveen.Palani “Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, ISO‑9001:2000 assessed delivery processes and provides solutions in areas of E-Business, ERP, Application Management Services, and EAI to customers in BFSI, Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global Delivery Model.” This e-mail and any attached files are confidential, proprietary, and may also be legally privileged information, and are intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient of this e-mail, please send it back to the person who sent it to you and delete the e-mail and any attached files and destroy any copies of it; you may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED] Quinnox Consultancy Services and/or any of its sister companies owns no responsibility for the views presented in the e-mail and any attached files unless the sender mentions so, with due authority of Quinnox Consultancy Services. Unauthorized reading, reproduction, publication, use, dissemination, forwarding, printing or copying of this e-mail and its attachments is prohibited. We have checked this message for any known viruses; however we decline any liability, in case of any damage caused by a non- detected virus. For more details about our company, visit http://www.Quinnox.com ___ -- 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] Dynamically change sip.conf properties.
Realtime only needs a sip reload if you are using static realtime, if you use the sippeers realtime it works just fine. See http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip Note that the settings change will only take effect when your client re-registers, so you may want to set a reasonably low qualify value. Simon Simon Elliston Ball [EMAIL PROTECTED] On 11 Dec 2007, at 15:15, asterisk wrote: I don't know of a way without reloading. Realtime still needs a sip reload. Look at the dial command. There are options that you can add that will disable re-invites per call. Doug Gillespie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Monday, December 10, 2007 12:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dynamically change sip.conf properties. Is there a way to dynamically alter the sip.conf properties of a SIP peer in runtime without doing a SIP reload? I am specifically thinking of enabling reinvites for users dynamically based on whether they are registered from a public address. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 ___ --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 on UML (User Mode Linux)
What's the current thinking on running Asterisk in a UML environment? I saw some discussion about Xen and asterisk on a Xen DomU. I'm currently running Asterisk in a UML and have noticed poorer quality on calls. I'm only using SIP and IAX2 trunks. No hardware adapters. I guess timing is important, but even if I could get the provider to install a kernel with the Zaptel Dummy timing device compiled in (impossible to install kernel modules in UML), I'm not convinced this would necessarily provide an accurate enough timing device. Is anyone else running their Asterisk instance in UML? If anyone is, what's the preferred way to keep timing accurate? Thinking I may have been too hasty in switching to UML... S. -- Simon Tennant ___ http://imaginator.com/~simon/contact signature.asc Description: OpenPGP digital signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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] Ringing sound doesn't work
Hi,
I have these extensions:
exten = 101,1,Dial(SIP/101,15)
exten = 102,1,Dial(SIP/102,15)
exten = 0,1,Dial(SIP/101SIP/102,15,r)
They work fine and I get the ringing sound if I dial them directly. However, I
also have this extension:
exten = s,1,Answer()
exten = s,2,Background(viagenie)
exten = s,3,WaitExten()
The ringing sound doesn't work for any extension if I use this one. I just get
silence until someone answers. How come?
I use Asterisk 1.4.10. I have attached my extensions.conf file to this email.
Thanks,
Simon
[globals]
SIPTRUNK=418555
IAXTRUNK=514555
[default]
exten = s,1,Answer()
exten = s,2,Background(viagenie)
exten = s,3,WaitExten()
exten = i,1,Background(invalid)
exten = i,n,Goto(s,1)
exten = t,1,Background(please-try-again)
exten = t,n,Goto(s,1)
[phones]
exten = 101,1,Dial(SIP/101,15)
exten = 101,n,Goto(201,1)
; Simon
exten = 102,1,Dial(SIP/102,15)
exten = 102,n,Voicemail(102)
exten = 201,n,Dial(SIP/[EMAIL PROTECTED],15)
exten = 201,n,Voicemail(101)
[ivr]
exten = 0,1,Dial(SIP/101SIP/102,15,r)
exten = 0,n,Goto(201,1)
exten = 8,1,Directory(default)
exten = #,1,Directory(default)
exten = 500,1,VoiceMailMain()
[voip_incoming]
exten = ${SIPTRUNK},1,Goto(s,1)
[voip_outgoing]
exten = _NXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _NXX,2,Dial(SIP/[EMAIL PROTECTED])
exten = _NXXNXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _NXXNXX,2,Dial(SIP/[EMAIL PROTECTED])
exten = _1NXXNXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _1NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9NXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _9NXX,2,Dial(SIP/418${EXTEN:[EMAIL PROTECTED])
exten = _9NXXNXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _9NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9.,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = N11,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = N11,2,Dial(SIP/[EMAIL PROTECTED])
[external]
include = default
include = phones
include = ivr
include = voip_incoming
[internal]
include = external
include = voip_outgoing
exten = 10,1,Goto(s,1)
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Re: [asterisk-users] Ringing sound doesn't work
On Wednesday 29 August 2007 10:46:18 Eric ManxPower Wieling wrote: You do not have a /etc/asterisk/indications.conf This file is used to provide ringing sounds AFTER a channel has been answered. Thanks a million times! ___ --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] Passing call duration to an AGI Script
Hi, What I did is first to dig a bit into the app_dial.c. I saw how the ANSWEREDTIME variable is created (end_time - answer_time). Then I added some lines to export the answer_time variable as a channel variable. I added these lines right after the answer_time decleration (line 1426 in ver 1.4.4) compiled and replaced the module. char toast2[80]; snprintf(toast2, sizeof(toast2), %ld, (long)(answer_time)); pbx_builtin_setvar_helper(chan, ANSWERTIME, toast2); This will put the call start time in unix timestamp in the channel variable ANSWERTIME. That's all. Hope it's helping. Adi. On 6/1/07, Luis Morales [EMAIL PROTECTED] wrote: Hi Adi, My be better if you send us the code about how did you do to catch and retrive the data from asterisk. Regards, Luis Morales On Fri, 2007-06-01 at 01:21 +0300, Adi Simon wrote: Hi Martin, Thanks for your reply. Maybe I wasn't clear enough. I am already running AGI periodically inside a call and it runs just fine. I'm using a patch for asterisk (can be found here) to do so. In short i'm using it for a prepaid system that needs to allow more than one prepaid call to run simultaneously. Anyway, I solved my problem by changing the code a bit. I added an AGI variable that holds the timestamp of the call answer time, thus allowing me to use it as an anchor for knowing how much time passed since the beginning of the call. Thanks again, Adi. On 5/31/07, Martin Smith [EMAIL PROTECTED] wrote: Hi Adi, AGI is probably best viewed like any other dialplan application (and with DeadAGI something that happens after, but anyway) -- in my opinion. I've seen people do some pretty wild stuff with it, but in the end, when I wonder if the Manager interface or AGI interface is most appropriate for a given task, I ask questions like Would I want to do this with another application? Is this even possible with another application?. In your case, I'd say you probably couldn't say... periodically execute a dialplan application that runs in the middle of a call without interrupting the call (with AGI, anyway). I'd recommend using the Manager interface and polling for call durations / listening for events and acting on the information you get back (I'd assume the answered duration is one of those values you could poll for). Hope this helps -- others, please jump in if I'm way wrong :) Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 __ From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Adi Simon Sent: Thursday, May 31, 2007 5:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Passing call duration to an AGI Script Hi, I'm trying to find a way of passing the actual call duration (something like ANSWEREDTIME) to an AGI script that runs periodically during a call. Any ideas? Thanks, Adi. ___ --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 -- .-.-.-.-.-.-.-.-.-.-.-.-.--.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-. Sigma Dental Plan Jefe de Soporte y Sistemas Telf. Oficina : +58(212)2646811 Cel.: +58(416)4242091 Caracas, Venezuela .-.-.-.-.-.-.-.-.-.-.-.-.--.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-. ___ --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] chan_iax2.so issues
Hi folks We've a few problems with a rebuild of one of our asterisk boxes, same kernel and configs as previously but unfortunately strange iax issues. If we load chan_iax2 then the system hits 100% CPU, if we do not load this module then all is well. I have tried removing the iax.conf and loading the chan_iax2 within the console and I got an error that included: iax2 show cache' already registered (or something close enough) This implies that another module is stepping on chan_iax2's toes. I've checked the loaded modules and none of them mention iax ... Has anyone else come across this issue or can shed some light on the module crossover. For reference the issue happens with both kernels I have tried 2.6.20.4 and 2.6.21.3 and both asterisk 1.4.2 and 1.4.4. Any help appreciated. Regards Simon Alman ___ --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] Passing call duration to an AGI Script
Hi, I'm trying to find a way of passing the actual call duration (something like ANSWEREDTIME) to an AGI script that runs periodically during a call. Any ideas? Thanks, Adi. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing call duration to an AGI Script
Hi Martin, Thanks for your reply. Maybe I wasn't clear enough. I am already running AGI periodically inside a call and it runs just fine. I'm using a patch for asterisk (can be found here http://asterisk-backports.org/wiki/index.php/User_talk:KNK) to do so. In short i'm using it for a prepaid system that needs to allow more than one prepaid call to run simultaneously. Anyway, I solved my problem by changing the code a bit. I added an AGI variable that holds the timestamp of the call answer time, thus allowing me to use it as an anchor for knowing how much time passed since the beginning of the call. Thanks again, Adi. On 5/31/07, Martin Smith [EMAIL PROTECTED] wrote: Hi Adi, AGI is probably best viewed like any other dialplan application (and with DeadAGI something that happens after, but anyway) -- in my opinion. I've seen people do some pretty wild stuff with it, but in the end, when I wonder if the Manager interface or AGI interface is most appropriate for a given task, I ask questions like Would I want to do this with another application? Is this even possible with another application?. In your case, I'd say you probably couldn't say... periodically execute a dialplan application that runs in the middle of a call without interrupting the call (with AGI, anyway). I'd recommend using the Manager interface and polling for call durations / listening for events and acting on the information you get back (I'd assume the answered duration is one of those values you could poll for). Hope this helps -- others, please jump in if I'm way wrong :) Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Adi Simon *Sent:* Thursday, May 31, 2007 5:54 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Passing call duration to an AGI Script Hi, I'm trying to find a way of passing the actual call duration (something like ANSWEREDTIME) to an AGI script that runs periodically during a call. Any ideas? Thanks, Adi. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940 no outgoing audio
Hi Salvatore The firmware is PS03-08-2-00. Unfortunately I can only packet capture on the asterisk server itself, but I am seeing: P 172.16.8.22 172.16.8.1: ICMP 172.16.8.22 udp port 2224 unreachable, length 36 IP 172.16.8.20 172.16.8.1: ICMP 172.16.8.20 udp port 17099 unreachable, length 36 IP 172.16.8.20 172.16.8.1: ICMP 172.16.8.20 udp port 17099 unreachable, length 36 IP 172.16.8.1 172.16.8.20: ICMP 172.16.8.1 udp port 17228 unreachable, length 208 Where x.1 is asterisk and x.20 is the cisco and x.22 is a polycom test phone. We also see these errors on our working network (asterisk 1.0.10) so they are possibly a red herring. I suspect RTP issues but am unsure how to proceed as the Cisco phones do not seem to allow rtp debugging via their console. For reference our rtp.conf is: [general] rtpstart=1 rtpend=2 rtpchecksums=yes (have tried with no and made no difference) Regards Simon Salvatore Giudice wrote: You should get a packet capture of both cisco-cisco and grandstream/polycom-cisco. Compare the SDP's. The cisco phone may not be able to understand the other vendor's devices. BTW, what version of firmware are you running on the cisco phones? -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Alman Sent: Tuesday, May 01, 2007 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7940 no outgoing audio Hi All We have a private network setup (no nat) with three types of phones connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco 7940 IP phones. When we ring polycom to grandstream or grandstream to polycom then both phones can send and receive voice fine and all is well. When we dial any combination of Cisco and either Polycom, or Granstream the Cisco, no voice is being sent but the Cisco can receive voice from the remote phone fine. When we dial Cisco to Cisco it all works fine. I am at a loss to figure this out and any help pointing me in the right direction would be appreciated. We are running an old Asterisk server with version 1.0.10 (yeah we know) and the same mix of hardware and configs works fine. On the new (problem) setup we are running Asterisk 1.4.2 and our Cisco firmware is 08-2-00. ___ --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] Cisco 7940 no outgoing audio
Hi All We have a private network setup (no nat) with three types of phones connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco 7940 IP phones. When we ring polycom to grandstream or grandstream to polycom then both phones can send and receive voice fine and all is well. When we dial any combination of Cisco and either Polycom, or Granstream the Cisco, no voice is being sent but the Cisco can receive voice from the remote phone fine. When we dial Cisco to Cisco it all works fine. I am at a loss to figure this out and any help pointing me in the right direction would be appreciated. We are running an old Asterisk server with version 1.0.10 (yeah we know) and the same mix of hardware and configs works fine. On the new (problem) setup we are running Asterisk 1.4.2 and our Cisco firmware is 08-2-00. Any help appreciated. Regards Simon Alman ___ --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] Debian asterisk-bristuff
I apologise now if I have managed to completely misunderstand this whole subject! I've built a small PC and loaded Etch 4.0 from the netinst cd. I did 'apt-get install asterisk-bristuff' which seemed to work but, it doesn't seem to have installed any files/modules for zaptel? ztcfg zaptel zaphfc I am using a billion hfc card Any pointers? -- Simon Faulkner 01538 303 900 Staffordshire Moorlands ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debian asterisk-bristuff
I am using a billion hfc card apt-get install zaptel-source m-a a-i zaptel Precompiled zaptel drivers should hopefully be added soon to Unstable / Testing . Thank you :-) ___ --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] Asterisk mini conference within IT360 in Toronto Apr30-May2nd
Hey all, The Toronto AUG has been working with Clue.ca and IT360 (LinuxWorld/NetworkWorld), and has put together a mini-asterisk conference within their larger conference: http://www.it360.ca/asterisk.cfm If you're interested, as an 'association' we get 25% off the listed prices. Our dicount code is: A101, and our association name is: Asterisk User Group. Early bird rates end Wednesday April 11th. Cheers, spd ___ --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] switchtype and signalling query
Hi Guys I'm configuring a TE212P card and have the following two entries in my /etc/asterisk/zapata.conf switchtype=dms100 signalling=pri_cpe When I reload asterisk I get the following messages: -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found [Mar 30 17:48:42] WARNING[2985]: chan_zap.c:11072 process_zap: Ignoring switchtype [Mar 30 17:48:42] WARNING[2985]: chan_zap.c:11072 process_zap: Ignoring signalling However pri show span 1 shows the right values set for both: ast1*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, In Alarm, Down, Active Switchtype: Nortel DMS100 Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 Should I be concerned as to the Warnings ? I'm not quite at the stage where I can test my setup yet and wanted to check before I get there. Many thanks for your time. Simon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] switchtype and signalling query
Cool, thanks for the info. Simon Doug Lytle wrote: Simon Alman wrote: Hi Guys I'm configuring a TE212P card and have the following two entries in my /etc/asterisk/zapata.conf switchtype=dms100 signalling=pri_cpe When I reload asterisk I get the following messages: -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found [Mar 30 17:48:42] WARNING[2985]: chan_zap.c:11072 process_zap: Ignoring switchtype [Mar 30 17:48:42] WARNING[2985]: chan_zap.c:11072 process_zap: Ignoring signalling On a reload, it is ignored since it is already set up. It's normal. Doug ___ --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] 3 way calling independent of phone hw.
I'm looking for a recipe for a 3 way call where one of the parties can (without using the flash button) dial-out and add a third participant to the call. I tried Googling but it seems I'm missing a key search term. The reason I wanted to avoid using the flash button is that some handsets don't have it (nokia E61 who's 2 way calling via sip is also broken) Something like: 1. party 1 calls party 2 2. either party 1 or 2 hits * on keypad 3. asterisk prompts for party 3's telephone number 4. asterisk dials party 3. 5. party 3 answers and is immediately added to 3-way call 6. the inviter has the option of pushing # to terminate party 3 (should the call only reach party 3's voicemail). Either that or a ways to do DISA from within the meet-me functionality. I can't imagine I'm the only person with this sort of requirement. -- Simon Tennant http://imaginator.com/~simon/contact signature.asc Description: OpenPGP digital signature ___ --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] Confederated SIP service.
'lo, A provider sets up an Asterisk box in order to service the needs of a small number of customers. The provider issues SIP handsets and the users register with sip.telco.com Thanks to the selection of a brilliant family of technologies, including SIP and Asterisk, the telco.com company grows and grows. Eventually, beyond the point that they can really hold all of the customer SIP registrations on one server. So, to improve scalability and redundancy, the provider installs four Asterisk servers to handle registrations. In a one server SIP environment, the dialplan is easy to setup exten = 1234,1,Dial(SIP/user1234,20,r) .. and if user1234 is registered, user1234 is dialed. But what about this multi-server environment? If the same extensions.conf line appears on all four asterisk servers, but the user is only registered to sip2.telco.com, how can the administrator make a Dial(SIP/user1234,20,r) on sip3.telco.comgo to the right user? Does a type of 'confederated sip registrations' system exist in Asterisk 1.4 ? Best regards, SJJD ___ --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] Asterisk not hanging up calls
I have noticed that Asterisk (version 1.2.13) is not hanging up a call when the wifi handset moves out of range. My setup is Nokia E61 connected to wifi access point (private IP range) and then to server on internet (public IP). I have been testing using the talking clock application, and walking out of range does not hang up the call. The call will continue for hours even though the handset re-registers on another access point 5 minutes later with a different public IP. Subsequent calls continue fine although I still see traffic heading out to the old public IP address of the wifi access point. I thought the SIP control channel would do some kind of keeping state and time out a call after x number of failed replies. I know there is a timeout option but would rather not set a timeout on all calls. Here's the sip.conf that I am using for the device: * Name : 105 Secret : Set MD5Secret: Not set Context : from-internal Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 101 VM Extension : asterisk LastMsgsSent : 2048 Call limit : 0 Dynamic : Yes Callerid : Simon Tennant (Nokia E61) Expire : 3595 Insecure : no Nat : Route ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : auto LastMsg : 0 ToHost : Addr-IP : 10.15.11.8 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 105 SIP Options : (none) Codecs : 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc) Codec Order : (alaw,ulaw,ilbc,speex,gsm,g729,g723) Status : OK (266 ms) Useragent: Reg. Contact : sip:[EMAIL PROTECTED] S. -- Simon Tennant http://imaginator.com/~simon/contact signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best inexpensive home office router for VoIP (QoS with maybe PoE)
We've had a lot of success with Thompson Speedtouch 780 routers, which have built in adsl modems, and two ATAs. They don't seem to use QoS in the strictest sense, but do a very good job of prioritising the traffic from their own ATAs. If you're happy to stick with analogue hansets instead of the SIP hardphones, they provide an excellent protection to upload bandwidth. They also seem to do some early dropping on incoming traffic to persuade the ISP's routers to slow down downloads once a call has been going for a bit, hence they can limit downloads as well. simon On 4 Jan 2007, at 17:56, Mike wrote: Hi, I'm looking for opinions on the best value router to use for home offices. It should work for a scenario in which there are 3 computers and 2 SIP phones, handling QoS so that the phones always have higher priority traffic than the PCs. (and not rely on the phones to do the QoS because some PCs may not be connected to the phones). QoS could be based on destination and source IP (i.e. an Asterisk server) or MAC address of the phones. Ideally with PoE, but at this point it's just a bonus. What are people on this list using? I've found that the mention QoS on a box doesn't guarantee any real QoS functionality. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel under FC6
The Fedora Extras rpm is tiny because it has nothing really of help in it. It's missing the modules. I've had some success on Fedora Core 6 using the ATrpms repository, which has the zaptel-kmdl package for most variations of kernels included in FC6. Simon On 14 Dec 2006, at 22:31, Yuan LIU wrote: From: Rudolf Ladyzhenskii [EMAIL PROTECTED] I guess, modules are mot there. Running find / -name zaptel* did not find any modules. Be careful here - wildcard expansion takes place locally unless you quote the string: $ find / -name 'zaptel*' Of course search from / is suboptimal as you are going to find your source as well, besides a looong search. I suggest starting from / lib/modules. Or do a simple ls. Seems that make is broken in some way. Rudolf ___ --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] WaitExten only reading 1 digit.
I am trying to setup an interactive menu where a caller hits the main
menu and can then dial an extension. As far as I can tell the
Waitexten app is failing to read 3 digits and just reading the first
and then announcing that it is invalid since all extensions are 3 digits.
How do I make Waitexten wait for 3 digits?
I have setup the extension 100 for users to reach the switchboard as
they would from outside:
[internal-extensions]
exten = 100,1,Goto(mainmenu,s,10)
exten = 101,1,Dial(SIP/101,30)
exten = 101,2,Voicemail(u101)
exten = 101,3,Hangup()
exten = 102,1,Dial(SIP/102,30)
exten = 102,2,Voicemail(u102)
exten = 102,3,Hangup()
dialing 100 then hits mainmenu
[mainmenu]
exten = s,10,Answer
exten = s,11,Wait(1)
exten = s,12,Background(buddy-cloud/welcome2)
exten = s,13,WaitExten(15)
exten = s,14,NoOp(Number dialed ${EXTEN})
include = internal-extensions
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
exten = i,1,Playback(invalid) ; That's not valid, try again
This is the output from me (x101) dialing the switchboard (x100)
-- Executing Goto(SIP/101-08186e70, mainmenu|s|10) in new stack
-- Goto (mainmenu,s,10)
-- Executing Answer(SIP/101-08186e70, ) in new stack
-- Executing Wait(SIP/101-08186e70, 1) in new stack
-- Executing BackGround(SIP/101-08186e70, buddy-cloud/welcome2)
in new stack
-- Playing 'buddy-cloud/welcome2' (language 'en')
-- Sent into invalid extension 's' in context 'mainmenu' on
SIP/101-08186e70
-- Executing Playback(SIP/101-08186e70, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Timeout on SIP/101-08186e70
== CDR updated on SIP/101-08186e70
-- Executing Playback(SIP/101-08186e70, vm-goodbye) in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing Hangup(SIP/101-08186e70, ) in new stack
== Spawn extension (mainmenu, t, 2) exited non-zero on 'SIP/101-08186e70'
Where am I going wrong and do I need to worry about Sent into invalid
extension 's' in context 'mainmenu' on SIP/101-08186e70 warnings?
S.
--
Simon Tennant http://imaginator.com/~simon/contact
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Re: [asterisk-users] WaitExten only reading 1 digit.
Doug Lytle wrote: Doug Lytle wrote: Simon Tennant wrote: [internal-extensions] exten = 100,1,Goto(mainmenu,s,10) You can't start at 10 on your menu, you have to start with 1. strange - I jumped into that context at 10 and numbered up from 10 - I thought that was ok. Also when I started numbering from 1 everything works. Cheers. -- Simon Tennant http://imaginator.com/~simon/contact signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK Colocation services
Hi,We can provide UK toll-free inbound and can recommend a co-lo provider where you'll have single-hop access to our network. Feel free to contact me off list.Kind regards,Simon On 9/27/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Can anyone direct me to a colo provider in the UKwhere I can park an asterisk server and buy UK toll free inbound services over SIP? Thanks Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER with multiple asterisk deployment
Mainly I have a problem of figuring out how to use them with dispatcher or any other mean of switching between asterisks. Do you have any configuration example of such? On 9/28/06, Simone Ricci [EMAIL PROTECTED] wrote: Adi Simon ha scritto: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). record_route() and loose_route() should help you, AFAIK. They don't?Cheers,Simone.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] SER with multiple asterisk deployment
Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). Please don't direct me to Asterisk+At+Large or the asterisk_integration page at openser.org as they are quite old and useless. What I seek are examples of ser.cfg or some advice from someone who actually managed to accomplish this. Thanks, Adi. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER with multiple asterisk deployment
Hi Zac, Thank you so much for your sincere answer. What you brought up is exactly what I encountered when I tried to find a solution for this, the documentation is inconsistent and ambiguous, and everywhere I look I end up with outdated examples that make little or no sense in the good case, or just don't compile due to being so old in the bad case. This is very frustrating but just by reading what you wrotewas very uplifting for me. Thanks again, Adi. On 9/27/06, Zac Amsler [EMAIL PROTECTED] wrote: Adi,It is possible to do what you are looking for. It is actually easy.There is a problem that I have found with ser/openser.. Documentation is difficult to read and some things are just not there, so you get peoplethat spend many hours trying to get these functions to work. In thesedays time is money, so the people that know how to do what you are seeking.. charge large amounts of money for a simple 50 line config file.I will tell you that everything you are looking for is documented inexamples. You will have to piece them together and make them work in harmony like the rest of us have.I suggest you look at voip user and piece the config together fromexamples there. It may also help you to read the source code of themodules that handle routing in ser. There are a few tricks that are hidden in the code.I am sorry for my vagueness. I am not able to share the configinformation due to an IP agreement with my company.(They think it is atrade secret)I wish you the best. Cheers,Zac Amsler, Network OperationsSur-Tel Communications, Inc. NetIQ Systems, LLC* US48, Canada, A-Z Wholesale Termination.* US48 Origination, Toll Free DIDs.* Toll Free Termination (FREE). Adi Simon wrote: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). Please don't direct me to Asterisk+At+Large http://www.voip-info.org/wiki-Asterisk+at+large or the asterisk_integration http://www.openser.org/dokuwiki/doku.php?id=asterisk_integration page at openser.org http://openser.org as they are quite old and useless. What I seek are examples of ser.cfg or some advice from someone who actually managed to accomplish this. Thanks, Adi. ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi Servers
Hi Doug,On 9/25/06, Douglas Garstang [EMAIL PROTECTED] wrote: Lets say you have a cluster of, say, 10 Asterisk servers. After doing a local lookup to see if a number is available locally, in order to find out if the number is available on one of the other 9 servers, this peer has to query all 9 remaining peers. Is that true?Yes, or it could send one query to a server which in turned queried the other 9. Either way though, all 9 get queried unless the answer was cached. Caching is tricky with registrations as you don't want to cache a registration which hasn't been renewed. Is there a way to have 'registration servers' that accept registrations from phones, and which somehow notify 'DUNDI servers' (two for redundancy) that the registration servers query? To terminate a call, a peer would only have to query the DUNDi servers, not every other peer. After looking at the config files, I can't imagine how this could work, or if it's even possible with DUNDi. Yes, it is possible to push peer information as well as pull it. You could also, as you say, limit the number of registration servers (i.e. servers doing both the registration and DUNDi) and then only query to them. I'm sure the hybrid model you suggest would work as well although it'd need testing to see whether you got more performance out of splitting the DUNDI and registration roles or just adding more dual-purpose machines. SimonDoug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Accounting and re-invite
Hi Ronald,On 9/19/06, Ronald Wiplinger [EMAIL PROTECTED] wrote: I am thinking if re-invite will interfere accounting.No it won't Please help me to figure it out:Phone A is registered at asterisk and calls a gateway. If the gatewayallows re-invite than the rtp would go directly from phone A to thegateway, while the sip messages are still going through Asterisk. Asterisk will be informed when the call ended.If it is a postpaid accounting, just bill the customer, however, how isit for a pre-paid (calling card user)?I think Asterisk will have no power to turn off the call from A to the gateway.Even more, if the gateway would allow to end a call and continue with anew call, the new call would not be billed (or would it)?The SIP messages control the call, irrespective of where the RTP goes so Asterisk can terminate/set-up calls exactly as if the RTP was being handled. I guess the solution must be re-invite=noHowever, re-invite=no means that each call is going with rtp also through my server, what means for a remote phone, I have to provide forboth legs the bandwidth.Personally, I would always handle the RTP on quality/accountability/consitency grounds but, yes, that will incur a bandwidth overhead. ___ --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] Zork Asterisk; zoip 0.2.0 released
ZoIP 0.2.0, the Zork/Asterisk bridge has finally been released. Now you too can play 80's era text adventures over the phone using text-to-speech, and speech recognition ;-) What's a text adventure like you ask? Well, depending on your skill, a typical dialog might go something like this: computer It is dark, you are likely to be eaten by a grue. me Turn on lantern computer You are in a cavern, axe marks line the wall. There is an angry troll here. me Kill troll with sword computer You swing, the troll dodges, and removes your head with his axe. You are dead. The INSTALL file is based on Ubuntu 6.06 LTS, though for the most part, you should be able to substitute 'apt-get' for 'yum'. See http://zoip.org for the goods, or download directly: http://demo.zoip.org/zoip-0.2.0.tar.gz Fairly major changes: - No longer need to run Festival as a service - No more DTMF, speech recognition using Sphinx2 - Bundled the necessary sphinx2 language model and acoustic model for the speech recognition. - It is now called as a standard AGI rather than EAGI - I've removed all the hardcoded paths - Added a configuration file For discussion, installation help, and such, see the forum linked off of http://zoip.org Cheers, spd | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | The Toronto Asterisk Users Group | Join the discussion group by visiting http://taug.ca | or by sending email to [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] macros in Realtime
Hi Ben,Yes it is but you need to remember to still include[macro-stdpbx1exten]switch = Realtime/in your extensions.confSimonOn 9/6/06,
Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello all,Another question related to Realtime.Is it possible to call macros using Realtime arch?I have a macro definition in table extensions_conf in my MySql db as: 30 | macro-stdpbx1exten | s |
1 | SET| fwdedNum=${DB(CFWD/${ARG1})}And calling the Macro using another entry in table extensions_conf inMysql db: 40 | pbx1 | _[345]. |
1 | Macro| stdpbx1exten|${EXTEN}I get errors like :Sep6 11:18:53 WARNING[14493]: app_macro.c:154 macro_exec: No suchcontext 'macro-stdpbx1exten' for macro 'stdpbx1exten'
Are there issues with the same??TiA-Ben.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
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Re: [asterisk-users] iax vs. sip?
Hi Steven,The provider's implementation will have a bigger affect than any differences within Asterisk, e.g. how they are load-balancing and whether in fact SIP is serviced by Asterisk at all. Compared like-for-like within Asterisk we find there is not a lot in it, with each having their own pros and cons. We support both and whilst we have more customers on SIP than IAX, currently favour IAX for new customers where they are undecided given lower support overhead and simplified load-balancing. I'd recommend you try both with the provider you're considering. Simonwww.esms.comOn 8/31/06, BerkHolz, Steven [EMAIL PROTECTED] wrote: I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIPmight bebetter to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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] detecting a users number using the dialplan or AGI
Hi All,I was hoping someone could help me with a problem I'm having determining a users number. Is there any way in the dialplan or with an AGI to detect what a users number is for use in a meetme conference? I am using the MeetMeAdmin function from within the dialplan.I would like one of my admins to be able to drop out of the conference and be able to kick the last user that joined the conference. I believe that I can do this using:MeetMeAdmin(confno|k|userno)keeping track of the confno is easy since I created it,but I don't know how to determine the user number of the last person that joined the conference. Is there a way to store this in a variable before they join the conference? Or perhaps a way to detect the last user to join the conferences number?Any help is appreciated.Cheers,- Simon Austin ___ --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] determining meetme user number
Hi,Is there a way to determine the MeetMeAdmin User number?I am using the MeetMeAdmin function from within the dialplan.I would like one of my admins to be able to drop out of the conference and be able to kick the last user that joined the conference. I believe that I can do this using:MeetMeAdmin(confno|k|userno)keeping track of the confno is easy since I created it,but I don't know how to determine the user number of the last person that joined the conference. Is there a way to store this in a variable before they join the conference? Or perhaps a way to detect the last user to join the conferences number?Cheers,Simon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Nokia E60/61/70 and SIP
Hi Benny,The E61 handles this just fine. With SIP as the default channel to dial and no WiFi coverage, you get a message asking if you'd like to dial by cellular. Works nicely other than a few stability issues. SimonOn 24 Aug 2006 11:23:39 +0200, Benny Amorsen [EMAIL PROTECTED] wrote: H == Haspers[EMAIL PROTECTED] writes:H We are using some E61 and E70's with asterisk. Only problem we haveH at this moment is that we are unable to use a password for the H authentication. I haven't found out yet why this isn't working.H They are working good, but I would like to see some small thingsH changed in future firmware versions (like being able to select H multiple WLAN points (Access groups) instead of just one.E70 works with passwords here. No trouble.The main issue is that the E70 can't automatically switch to cellularwhen the phone is out of WLAN coverage. It is a bit silly to have to click option-s___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Nokia E60/61/70 and SIP
Hi Haspers,Makes sure you have created an 'Internet tel' profile. It doesn't appear to do anything but was vital in getting it working for me. The other settings in the how to look sensible.Simon On 8/24/06, Haspers [EMAIL PROTECTED] wrote: Strange,What settings do you use? I followed this linkhttp://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html but without anyluck. -Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of Benny AmorsenSent: donderdag 24 augustus 2006 11:24To: asterisk-users@lists.digium.comSubject: [asterisk-users] Re: Nokia E60/61/70 and SIP H == Haspers[EMAIL PROTECTED] writes:H We are using some E61 and E70's with asterisk. Only problem we haveH at this moment is that we are unable to use a password for the H authentication. I haven't found out yet why this isn't working.H They are working good, but I would like to see some small thingsH changed in future firmware versions (like being able to select H multiple WLAN points (Access groups) instead of just one.E70 works with passwords here. No trouble.The main issue is that the E70 can't automatically switch to cellular whenthe phone is out of WLAN coverage. It is a bit silly to have to click option-s___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E61
Hi Andreas,That is incorrect. It works just fine through NAT providing:- The server is proxying RTP as it has no support for STUN etc.- The NAT is the basic domestic router style, not a full blown firewall requiring port mappings SimonOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED] wrote: Anyone here use the Nokia E61 ? I am looking to invest in a wifi phone and I want to get the best. Is it good as far as reception ? That is of most importance to me. Thanks.I've tried it in the last couple of days. The biggest issue for me ist that it HAS to be on the same side of a NAT as theserver it talks to (asterisk, ser, etc). If it is on theprivate side of a NAT and the server is on the public side, itdoesn't work. I've read something on the Nokia forums that Nokia is aware of the problem and it will be solved.My problem is that they want to solve this using STUN etc,while I would prefer they also wouldn't have the softwarecare if it is on the inside of a NAT like most other CPE's so our platform can take care of things.--Andreas SikkemaBBeyondSoftware EngineerPlaneetbaan 4+31 (0)23 70743422132 HZ Hoofddorp___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E61
Hi Andreas,I'm on 1.0610.04.04 19-04-06 RM-89WOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED] wrote:Simon, That is incorrect. It works just fine through NAT providing: - The server is proxying RTP as it has no support for STUN etc. - The NAT is the basic domestic router style, not a full blown firewall requiring port mappingsStrange, then you must have some other firmware, because I just can't get it registered at all, let alone make calls.We do have proxies for RTP ;-)--Andreas SikkemaBBeyondSoftware EngineerPlaneetbaan 4+31 (0)23 70743422132 HZ Hoofddorp ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Nokia E60/61/70 and SIP
Hi,Yes, the proxy and registrar settings are identical in my set-up. Getting registered is the hard part but you've done that. I'd be looking at the Asterisk end of things rather than the E61 to see why that authentication issue is arising. WOn 8/24/06, Haspers [EMAIL PROTECTED] wrote: I've got them all. It registers correctly with Asterisk, and get incoming calls, but it complaints about outgoing calls (Connection Error). SIP Debug is giving me: SIP/2.0 407 Proxy Authentication Required But those settings are the same (Proxy Server/Registrar Server). So what could be the problem? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Simon WoodheadSent: donderdag 24 augustus 2006 12:51To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Re: Nokia E60/61/70 and SIP Hi Haspers,Makes sure you have created an 'Internet tel' profile. It doesn't appear to do anything but was vital in getting it working for me. The other settings in the how to look sensible.Simon On 8/24/06, Haspers [EMAIL PROTECTED] wrote: Strange,What settings do you use? I followed this linkhttp://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html but without anyluck. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Benny AmorsenSent: donderdag 24 augustus 2006 11:24To: asterisk-users@lists.digium.comSubject: [asterisk-users] Re: Nokia E60/61/70 and SIP H == Haspers[EMAIL PROTECTED] writes:H We are using some E61 and E70's with asterisk. Only problem we haveH at this moment is that we are unable to use a password for the H authentication. I haven't found out yet why this isn't working.H They are working good, but I would like to see some small thingsH changed in future firmware versions (like being able to selectH multiple WLAN points (Access groups) instead of just one.E70 works with passwords here. No trouble.The main issue is that the E70 can't automatically switch to cellular whenthe phone is out of WLAN coverage. It is a bit silly to have to click option-s___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Apache for FastAGI
Hi Doug,We run with SOAP on both the client/server side, i.e. conventional AGI making SOAP request to SOAP server running on Apache. I'm sure FastAGI would be quicker but wasn't around and other than the overhead of making the SOAP request is very lightweight on the Asterisk side. SimonOn 8/22/06, Douglas Garstang [EMAIL PROTECTED] wrote: I'm not sure how one would build a HTTP header on the client side, given that all you have to work with is a single line entry in extensions.conf. -Original Message- From: Tielin Xu [mailto: [EMAIL PROTECTED]] Sent: Friday, August 18, 2006 12:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Apache for FastAGI It is an valid option, but you have to build a HTTP header in your request to your web server, which CGI programs or Java servlets on web server could interpret your request from Asterisk. Tielin [EMAIL PROTECTED] 08/18/06 11:28 AM Here's an idea... Rather than writing your own multi-thread socket server for use with FastAGI, has anyone tried to use an Apache web server instead? After all, it does all that for you. I just gave it a shot, but Asterisk tries to send all the agi params to the web server, which it doesn't like it... [Fri Aug 18 12:25:28 2006] [error] [client xxx.yyy.141.162] Invalid URI in request agi_network: yes Doug ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls over VPN
We've done this with OpenVPN and it works fine. I'd recommend that the VPN server is not on the same box as Asterisk. Stick it on a firewall/gateway box giving access to the network containing the Asterisk boxes behind it. This way the Asterisk box(es) is seeing normal unencrypted traffic and the VPN server(s) can be specified to meet the VPN requirement. On 8/23/06, Joseph [EMAIL PROTECTED] wrote: Is anybody making calls over VPN?If so what is the penalty asencryption is involved.I was planning to use VPN to register Sipura units to my local asteriskthis way I don't have to deal with NAT issues. --#Joseph___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Extensions -- Comments?
note http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions comment from Philipp Dunkel. On 22 Aug 2006, at 17:13, Douglas Garstang wrote: -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Extensions -- Comments? Douglas Garstang wrote: Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? Yes, DON'T USE REALTIME! I wish it was that easy. We started looking at realtime again, because the option of building the config files with a script querying the database became daunting. It doesn't matter where you turn in Asterisk, there's gotcha's. For example, you can't put the hint stuff into realtime, and there's no inherint way to comment extensions. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Extensions -- Comments?
I use a view as the extensions table allowing you to add flags to your source table which can be filtered out in the view. The view also allows me to store users in an easier to handle way for our web app (eg, a users/extension numbers table, device table, phone models table for default sip settings) which are then joined together in various ways to produce views for the extensions and a sipdevices. Simon On 22 Aug 2006, at 15:20, Douglas Garstang wrote: The unofficial docs on the voip wiki for the realtime extensions table structure is: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Extensions -- Comments?
no it doesn't. you could just change the context field for the extensions you wanted to comment out. On 22 Aug 2006, at 16:11, Douglas Garstang wrote: Thanks, but that means I'd have to effectively comment out every extension in that context, which isn't very feesible. -Original Message- From: Joe Dennick [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 22, 2006 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Extensions -- Comments? Actually, you could just change to context. If a context isn't established in the extensions.conf file, it won't every be included in the dial-plan. As such, if you change the context on some realtime entries, they won't be included in the dial-plan. Later, all you have to do is change the context back with a simple SQL UPDATE statement. Douglas Garstang wrote: The unofficial docs on the voip wiki for the realtime extensions table structure is: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? Doug. ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel install - Fedora Core 5
I managed to get zaptel to compile reasonably easily on 2.6.17-1.2157_FC5smp. However, the yum repo sites do not provide devel packages for 2.6.17-2174 for some reason last time I checked, hence couldn't get it to build on that kernel. You could probably create the devel package without too much trouble from the srpm, but it's a lot easier to stick to 2157. If anyone else have managed to get FC5 to install the correct devel packages for the latest kernel, please let me know! Simon On 21 Aug 2006, at 11:52, Tomislav Parčina wrote: I'm trying to install Zaptel 1.2.7 on Fedora Core 5 with 2.6.17-1.2174_FC5 kernel from source code. When I untar Zaptel and execute this is error that I get. cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE- DSTANDALONE_ZAPATA -DZAPTE L_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE- DSTANDALONE_ZAPATA -DZAPTE L_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.17-1.2174_FC5/build You do not appear to have the sources for the 2.6.17-1.2174_FC5 kernel installed . make: *** [linux26] Error 1 What could be the problem? How to solve it? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Zaptel install - Fedora Core 5
In which case your best bet is probably to install with an rpm -- rebuilt on the source rpm. simon On 21 Aug 2006, at 12:36, Tomislav Parčina wrote: In article 344F8B3D-6591-4001-9DE6- [EMAIL PROTECTED], [EMAIL PROTECTED] says... I managed to get zaptel to compile reasonably easily on 2.6.17-1.2157_FC5smp. However, the yum repo sites do not provide devel packages for 2.6.17-2174 for some reason last time I checked, hence couldn't get it to build on that kernel. You could probably create the devel package without too much trouble from the srpm, but it's a lot easier to stick to 2157. Hi Simon! I have to use 2.6.17-1.2157 because I have precompiled vt1211 chip (sensors for VIA motherboards) driver for that kernel. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] VoiceMail and Fax on same extension
Hi, I'm trying to accomplish having a single extension that always answers with an automated voicemail prompt and record a user message, but can recognize if the call is fax and handle it accordingly. Anyone here has any experience with this kind of configuration? Thanks, Adi. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX unstable with large number of calls?
Hi Curt,That probably suggests that with SIP they're handing off the RTP to their upstream provider and just dealing with the signalling which is very low overhead. With IAX they have to transport both unless they're interconnecting upstream by IAX and can transfer. In my experience the load is about the same for IAX or SIP+RTP and limited [on a single box] by the specification of the box itself. Whilst I risk being shot down, I'd be wary of any provider who isn't themselves handling the RTP for a multitude of quality reasons (and just because that is what you're paying them for), as well as one who quotes capacities in single box terms. SimonOn 8/15/06, Curt Shaffer [EMAIL PROTECTED] wrote: I was just talking with an unnamed provider and the guy told me that they recommend their users not to use IAX because it is unstable at 50 concurrent calls and unusable at 100 or more calls. Now I have personally worked on an asterisk box that was pushing more than 50 and there were no problems. Anyone else out there have any data either for or against this suggestion? Thanks Curt ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] accessing dialplan global variables in agi
Russel, I did see your note. Thanks for the patch. I haven't had a chance to apply it yet. I hope to apply it tommorow. I'll let you know the results as soon as possible. Thanks for your quick response. That was the fastest response to a bug fix request I've ever seen. Cheers, - SimonOn 7/29/06, Russell Bryant [EMAIL PROTECTED] wrote: - Simon Austin [EMAIL PROTECTED] wrote: I have confirmed that GET VARIABLE doesn't return global variables in version 1.2.10 and submitted the following bug report: http://bugs.digium.com/view.php?id=7609I'm not sure if you have seen it, but I posted a patch to your bug report about an hour after you reported it that should fix the issue.Let me know what happens. Thanks,--Russell BryantSoftware DeveloperDigium, Inc.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] playing a sound into a meetme conf
Thank you.I had previously seen the Local channel channel but I didn't completely understand how it worked. That was a really good explanation of how I can do it.Cheers,- Simon
On 7/27/06, Moises Silva [EMAIL PROTECTED] wrote:
may be im missing something, but i think the pseudo channel you arelooking for is called Local and you can call some extension that youknow the only thing it does is play the message you need. So you can
originate a call to that Local channel and bridge it to the Meetmeconference where your users are waiting.[meetme-play-message]exten = s,1,Answer()exten = s,2,Playback(were_sorry_the_translator_is_gone)
exten = s,3,Hangup()[receive-meetme-message]exten = s,1,Answer()exten = s,2,AGI(some.pl) /* assuming is needed to execute agi to knowwich conference to join*/exten = s,3,Meetme(${variable_conference_set_from_perl_agi})
exten = s,4,Hangup()then, when a translator is gone, from the DeadAgi execute a manageraction Originate to call Local channelAction: OriginateChannel: Local/[EMAIL PROTECTED]
Context: receive-meetme-messageextension: spriority: 1You can do it in both directions :)It would be enough?RegardsOn 7/27/06, Simon Austin
[EMAIL PROTECTED] wrote: Hi All, I have a problem and I'm not sure if a solution is possible without using the asterisk testing code. I am developing a volunteer translation service that users can dial into.
I have a list of volunteer translators cell phone numbers stored in a mysql database along with times that they have volunteered to act as translators.That I pull from using some perl AGI scripts.
A user calls, I ask which language they need help/translation with, then I put the users into a meetme conference while I call translators and play them a message asking if they're available at this time.They can refuse
or accept the call. Once I get a translator that has accepted the call I connect the translator as an administrator to the meetme conference that is holding the user that is listening to music on hold.
That is all working quite well with the Dialplan and AGI scripts I have set up. Problems happen when the translator drops the call midway through the conversation.i.e. Losing cell phone service.
When that happens I need a way to play a message to the user to let them know that the translator has been lost and we're looking for a new one. I then need to put back the music on hold, then run deadagi scripts to
find a new translator to connect to the meetme conference to help out the user. What is currently happening is that the user is left in the conference alone forever listening to MOH.
I think there are two ways to do this, but I can't find out how to do either from any documentation I've found. 1. Break the user out of the meetme conf and back into the dialplan.
- If I kick them from the conference they are immediately hung up on and I don't know how to stop this from happening. - There is function that is available in Asterisk 1.4 called ManagerRedirect that seems like it could do this for me, but i'd rather
not try to integrate this into 1.2.10 because I fear breaking too many other things and running 1.4 (testing) just isn't an option at this time. (details here:
http://bugs.digium.com/view.php?id=6508) 2. Play a message into the conference - Can I join a new pseudo channel that I've created to a meetme conf that plays a message?Does anyone know how to do this?
- Can I override the MOH and stream a recorded message into the conference with only the single user in the meetme conf? Any help/ideas are appreciated. Cheers,
- Simon Simon Austin http://simon.openflows.org ___
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Re: [asterisk-users] accessing dialplan global variables in agi
I have confirmed that GET VARIABLE doesn't return global variables in version 1.2.10 and submitted the following bug report: http://bugs.digium.com/view.php?id=7609 Cheers,On 7/27/06, Russell Bryant [EMAIL PROTECTED] wrote: On Thu, 2006-07-27 at 19:02 -0400, Simon Austin wrote: Is it possible to access dialplan global variables from the AGI?It certainly should be. voip-info.org indicates that the GET VARIABLE (http://www.voip-info.org/wiki/view/get+variable) command can't get them.If you try it out and this does not work, I would consider that a bug. Feel free to report it on bugs.digium.com if that is the case.--Russell BryantSoftware DeveloperDigium, Inc.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] accessing dialplan global variables in agi
I first tried using the perl AGI libraries, then when that didn't work I tried using GET VARIABLE directly.The global variables I'm talking about are the globals that are defined in the dialplan under [globals]. Not the predefined channel variables ( e.g. CALLERID)I confirmed that there was not something wrong with my code by correctly retrieving both som predefined channel variables and some local variables that I set using Set().Can you please confirm that you're able to retrieve global variables set in the [globals] section of the dialplan? Cheers,- SimonOn 7/28/06, Don [EMAIL PROTECTED] wrote: Worked on same version when I did it...using PHP - Original Message - From: Simon Austin To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, July 28, 2006 3:52 PM Subject: Re: [asterisk-users] accessing dialplan global variables in agi I have confirmed that GET VARIABLE doesn't return global variables in version 1.2.10 and submitted the following bug report: http://bugs.digium.com/view.php?id=7609 Cheers, On 7/27/06, Russell Bryant [EMAIL PROTECTED] wrote: On Thu, 2006-07-27 at 19:02 -0400, Simon Austin wrote: Is it possible to access dialplan global variables from the AGI?It certainly should be. voip-info.org indicates that the GET VARIABLE (http://www.voip-info.org/wiki/view/get+variable) command can't get them.If you try it out and this does not work, I would consider that a bug. Feel free to report it on bugs.digium.com if that is the case.--Russell BryantSoftware DeveloperDigium, Inc.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.10.4/401 - Release Date: 7/26/2006 ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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] playing a sound into a meetme conf
Hi All,I have a problem and I'm not sure if a solution is possible without using the asterisk testing code.I am developing a volunteer translation service that users can dial into. I have a list of volunteer translators cell phone numbers stored in a mysql database along with times that they have volunteered to act as translators. That I pull from using some perl AGI scripts. A user calls, I ask which language they need help/translation with, then I put the users into a meetme conference while I call translators and play them a message asking if they're available at this time. They can refuse or accept the call. Once I get a translator that has accepted the call I connect the translator as an administrator to the meetme conference that is holding the user that is listening to music on hold.That is all working quite well with the Dialplan and AGI scripts I have set up. Problems happen when the translator drops the call midway through the conversation. i.e. Losing cell phone service.When that happens I need a way to play a message to the user to let them know that the translator has been lost and we're looking for a new one. I then need to put back the music on hold, then run deadagi scripts to find a new translator to connect to the meetme conference to help out the user.What is currently happening is that the user is left in the conference alone forever listening to MOH. I think there are two ways to do this, but I can't find out how to do either from any documentation I've found.1. Break the user out of the meetme conf and back into the dialplan. - If I kick them from the conference they are immediately hung up on and I don't know how to stop this from happening. - There is function that is available in Asterisk 1.4 called ManagerRedirect that seems like it could do this for me, but i'd rather not try to integrate this into 1.2.10 because I fear breaking too many other things and running 1.4 (testing) just isn't an option at this time.(details here: http://bugs.digium.com/view.php?id=6508)2. Play a message into the conference - Can I join a new pseudo channel that I've created to a meetme conf that plays a message? Does anyone know how to do this? - Can I override the MOH and stream a recorded message into the conference with only the single user in the meetme conf?Any help/ideas are appreciated.Cheers, - Simon ___ --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
