Re: [asterisk-users] All the four lights blinking
Thank you very much Kevin P. Fleming Christian
It worked for me :)
For information . I am using these settings given below
After settings then in CLI
console dial 1...@test_out
if you plan test the board you can try make a E1/T1 cable
https://www.juniper.net/techpubs/hardware/m40/m40-hwguide/html/pinout4.htmlhttp://www.linkedin.com/redirect?url=https%3A%2F%2Fwww%2Ejuniper%2Enet%2Ftechpubs%2Fhardware%2Fm40%2Fm40-hwguide%2Fhtml%2Fpinout4%2Ehtmlurlhash=Xhw3_t=tracking_disc
and connect two E1/T1 on the same board.
In system.conf
span=1,0,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
span=2,1,0,ccs,hdb3,crc4
# termtype: te
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
In chan_dahdi.conf
[channels]
callerid=asreceived
resetinterval=never
pridialplan=unknown
signalling=pri_cpe
switchtype=euroisdn
immediate=yes
group=0
context=test_in
channel=1-15,17-31
group=1
signalling=pri_net
context=test_out
channel=32-46,48-62
And in extensions.conf
[test_in]
exten = s,1,Answer();
exten = s,n,Playback(tt-monkeys)
exten = s,n,hangup();
[test_out]
exten = _X.,1,dial(DAHDI/g1/${EXTEN},60,R)
On Sat, Sep 12, 2009 at 3:23 AM, Christian Victor
christ...@victormedia.dewrote:
2009/9/11 ABBAS SHAKEEL shakeel.abbas@gmail.com
Thanks you very much Kevin.I will try it by connecting one end of
Ethernet cable to one slot and other to second slot . Configuring one
as pri_net and the other as pri_cpe.
I will provide you feed on monday either i succed or not
Remember that you CANT NOT use an Ethernet cross-over cable. You need to
get a E1 cross-over cable. Google for the pinout.
Christian
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Best Regards
Shakeel Abbas
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[asterisk-users] user=phone
Hi, How can I add to the from header ;phone=user ? I have set in the sip.conf *usereqphone* = yes, but it still not appears in the from header. Thanks Szasz Szabolcs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G.729 for Asterisk
Hello I have a confusion relating to G.729 codec. I know how to install where to get license but i really don't know why we need it? Why people use G.729 codec with asterisk? look all functionality can be done with out it ie calling from sip to iax protocol and sip/ iax to E1, then why we need this?? regards Adam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfers security
2009/9/9 Stanisław Pitucha s...@gradwell.net: I've got different customers that may use the same asterisk. Each user can blind-transfer a call to whatever place they want. But of course the transferring side should be billed for it. What can I do to see the difference between the channels here? Trying again, since I didn't get any responses... but someone has to know the answer ;) I know I can get the channel name via BLINDTRANSFER, but that doesn't really help me. What I need is either the CALLERID(num) if the calling side initiated the transfer, or either EXTEN or some other custom variable from the calling leg if it was the called party that does the transfer. Has anyone solved this billing problem in any way? (well - any apart from the asterisk-generated cdr-s - I don't really want to start relying on them) Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfers security
On 14/09/09 10:05 PM, Stanisław Pitucha wrote: 2009/9/9 Stanisław Pituchas...@gradwell.net: I've got different customers that may use the same asterisk. Each user can blind-transfer a call to whatever place they want. But of course the transferring side should be billed for it. What can I do to see the difference between the channels here? Trying again, since I didn't get any responses... but someone has to know the answer ;) I know I can get the channel name via BLINDTRANSFER, but that doesn't really help me. What I need is either the CALLERID(num) if the calling side initiated the transfer, or either EXTEN or some other custom variable from the calling leg if it was the called party that does the transfer. Has anyone solved this billing problem in any way? (well - any apart from the asterisk-generated cdr-s - I don't really want to start relying on them) For every billable item we use a code for the account and store it in... accountcode :) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Something with Dahdi and reversal event...
I don't know what the following means : [Sep 14 10:41:25] DEBUG[4438] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 3, state 6 [Sep 14 10:41:25] DEBUG[4438] chan_dahdi.c: Polarity Reversal event occured - DEBUG 1: channel 3, state 6, pol= 1, aonp= 0, honp= 1, pdelay= 600, tv= 294924 [Sep 14 10:41:25] DEBUG[4438] chan_dahdi.c: Polarity Reversal detected and now Hanging up on channel 3 [Sep 14 10:41:25] DEBUG[4438] chan_dahdi.c: Polarity Reversal event occured - DEBUG 2: channel 3, state 6, pol= 0, aonp= 0, honp= 1, pdelay= 600, tv= 294924 [Sep 14 10:41:25] VERBOSE[4438] logger.c: -- Executing [...@open:1] NoOp(DAHDI/3-1, extensie hangup - tel2268191 - winkel open) in new stack [Sep 14 10:41:25] VERBOSE[4438] logger.c: -- Executing [...@open:2] NoOp(DAHDI/3-1, hangup-cause : 16) in new stack Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 for Asterisk
silent sayz wrote: why we need it? IMHO, Actually we don't. Why people use G.729 codec with asterisk? Because it has a very good bandwidth/quality relation. Or because you need to inter-operate with another system based on this codec. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfers security
2009/9/14 Matt Riddell li...@venturevoip.com: For every billable item we use a code for the account and store it in... accountcode :) I'm not sure that actually answers my question... If you have a A-B call and set accountcode for A on it, then B does a blind transfer, how do you set the correct accountcode then? (assuming B is a different customer and blind-transfers you to pstn) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Reporting
Hi Folks, sorry for the delay ... I found that the documentation was rather iffy .. I finally found the defines.php in the lib subdirectory and figured out how to give the MySQL port with the host and it all works fine now. Gary Baribault Courriel: g...@baribault.net GPG Key: 0xFA812835 GPG Fingerprint: 8597 4D3D 3C3D 4247 077C 9FF9 E412 CAC4 FA81 2835 On 09/11/2009 01:09 AM, Matt Riddell wrote: On 11/09/09 7:11 AM, Gary Baribault wrote: Hi all, I'm looking for a reporting solution for Asterisk CDRs. I have a small Asterisk server that will eventually have 4 - 6 trunks. the system is up and the CDRs are being written to a MySQL DB. I tried installing Areski, but had no success .. I assume it's no longer supported... the last update was in March 2005 according to this page.. http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Has anyone got that it running? My server is OpenSuSE 11.2 with Apache 2 and PHP5, which is probably the problem.. the software probably needs PHP4. Yeah we use it from time to time. What do you mean it wasn't working? Did you get some errors or something? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 for Asterisk
A codec is completely independent from the protocol used. I guess you're mistaking the protocol (SIP, IAX2, H.323) for the codec (G.711, G.729, GSM, iLBC). You don't have to necessarily use G.729 codec, but a lot of VoIP providers use just G.729 or G.711 codecs in their platforms, so it's a matter of interoperability. Also, IMHO, G.729 has the best voice quality for a compressed codec. Many users can't notice the difference in a call using G.711 and G.729. The same can not always be said for other compressed codecs like GSM or iLBC. Vinícius Fontes www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP - silent sayz silent.s...@gmail.com escreveu: Hello I have a confusion relating to G.729 codec. I know how to install where to get license but i really don't know why we need it? Why people use G.729 codec with asterisk? look all functionality can be done with out it ie calling from sip to iax protocol and sip/ iax to E1, then why we need this?? regards Adam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfers security
2009/9/14 Olle E. Johansson o...@edvina.net: Make sure that each device has a TRANSFER_CONTEXT dialplan variable. What about a situation where sip devices register at a proxy in front of many asterisks and asterisks authorise all calls from that proxy? I.e. I don't have any devices that asterisk would know about. That way as far as asterisk is concerned, the call is a simple trunk call and the B side (in A-B call) doesn't trigger any TRANSFER_CONTEXT setting when doing a transfer. I hacked together a solution that works for me now, but I'd rather solve this problem properly. My solution was that the A-B call gets out to the device via rB context. When A does a transfer current.chan1 (in handle_refer) has CALLERID(num) set to rB. When B transfers, callerid is obviously A. So I just copy that value to some variable in the new channel and bill based on that in a common transfer context. Still - I'd rather find a solution that doesn't involve patching chan_sip... (and doesn't require me to set up sip users on all asterisks). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfers security
14 sep 2009 kl. 12.05 skrev Stanisław Pitucha: 2009/9/9 Stanisław Pitucha s...@gradwell.net: I've got different customers that may use the same asterisk. Each user can blind-transfer a call to whatever place they want. But of course the transferring side should be billed for it. What can I do to see the difference between the channels here? Trying again, since I didn't get any responses... but someone has to know the answer ;) I know I can get the channel name via BLINDTRANSFER, but that doesn't really help me. What I need is either the CALLERID(num) if the calling side initiated the transfer, or either EXTEN or some other custom variable from the calling leg if it was the called party that does the transfer. Has anyone solved this billing problem in any way? (well - any apart from the asterisk-generated cdr-s - I don't really want to start relying on them) Make sure that each device has a TRANSFER_CONTEXT dialplan variable. That way, you can send different customers to different TRANSFER_CONTEXTs. Only transfers end up in these contexts, so you can check who's doing what and set accountcodes, reset cdrs, play prompts and have fun. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstriCon 2009: 30 days, 5 reasons discount code
We're down to slightly less than a month between now and AstriCon! October 13-15 is drawing close. If you've not booked your travel reservations to Phoenix, now is the time to do it! Sept 23rd is the cutoff date for the room discount, and we've requested another block of rooms for attendees. The Renaissance is the exclusive hotel of AstriCon 2009. It is where most of the attendees and exhibitors stay, and is a great place for networking! Please make sure you book your hotel rooms right now to ensure you get the AstriCon fantastic discounted room rate. Availability is on a first-come first-served basis and the cut off date for the special rate is September 23 2009. Renaissance Glendale Resort Spa http://cwp.marriott.com/phxgr/astricon09/ Special rate - $144/night There are other hotels in the area, so if the hotel fills (as it did last year) there are other options. But getting a room at the Renaissance is probably your best bet, since you won't have to trudge across the arid parking lots or drive to another venue. There are lots of restaurants close-by in the new entertainment center - I didn't go to the same place twice last year! To answer the question that seems to be on everyone's lips: yes, AstriCon looks to be as big than last year, if not significantly bigger. I know the economic situation is weighing on everyone's mind, but Open Source Asterisk installations are up and what is hurting the big guys is putting some wind under our wings. AstriCon is where you'll see lots of people who are winning deals, creating revenue, and building the market of the PBX that is now the most-installed platform in North America (we're hoping to say world-wide VERY soon.) Now, to encourage having everyone book a TINY bit in advance instead of all at the last minute (who, you? book at the last minute? I know I'm not talking about you.) I'll again announce that we have a discount code that you can use on your sign-up, which will give you a 15% break on the conference price. The code is AC09 and you'd enter it on the registration page (http://www.astricon.net/ attendRegister.aspx) to get your discount. Top 5 Reasons to attend AstriCon: * The talks! This is yet another stellar line-up of talks this year, with a wide array of fascinating examples of how Asterisk is being used to solve novel problems. How can you make your business or project more profitable and effective? These talks focus on those questions, and more. The sub-tracks on cloud computing and government/ large enterprise implementations are creating quite a bit of interest this year, and the speakers have extensive practical advice to dispense on all topic areas. * Trade information - Open Source isn't just the software. To a large degree, our user and development community members cooperate with each other to solve all kinds of problems. Ask others about their experiences, and offer your own in the informal setting around the conference. The market around Open Source software doesn't just have code as its only currency! The conference is for information exchange, and this just might be the most valuable thing you take home. * Vendor area - check out the new technologies from hardware vendors, software vendors, and service providers! The market changes; make sure you know what the most current methods and products are. * Put names to faces - that person you've been talking with for a year on IRC or IM but have never met? That consultant whose emails you've been reading on the mailing list? That customer you've been trying to get pay attention to you? Chances are good they'll be at AstriCon, and having that face-to-face conversation is sometimes the trigger you need to get a project going. * Meet entirely new people - the best experiences at AstriCon come from the most unexpected places. That person you sit beside in a talk, the lunch table you share with others, the person in the elevator with you - the interactions you have will expose you to new people whose projects will amaze and interest you, and possibly even lead to your changing your methods or finding new business. I really hope I see you there! JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722 problems with IAX
2009/9/9 Armin Schindler ar...@melware.de No, I didn't miss that. See my text. I mentioned this because I think this might be the reason of the problem and the incorrect handling in jitterbuffer, if it is the jitterbuffer. It is just a guess, since everything else seems to work good. The question is why does G.722 via IAX has problems. Is anyone using it and can say it works in his setup? Hi, I'm not sure if Steve Kann is still around the project, but if not, I'm familiar with chan_iax2.c and mostly familiar with the iax2 jitter buffer so I might have a go at fixing the problem. Will you open a bug on the bugs.digium.com bug tracker. I did do a test from a SNOM820 (yum) via an IAX trunk with jitter buffer and got the same nasty jerky audio. This is a recent checkout of branch-1.4. Regards, Steve Davies ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI Dial 9 Receiving Setup Acknowledge
I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make
calls from the Toshiba to Asterisk and internal calls from Asterisk to
the Toshiba. What I can't do is make an call with an outside
destination from Asterisk to the Toshiba. The Toshiba is looking for 9
to grab an outside line then it expects to see the 10 digits. In the
FreePBX dial plan I use 9|. which sends 9 plus the 10 digit number.
Using Wireshark to look at the QSIG commands coming from a Sangoma
wanpipemon trace I see the following for an Asterisk to Toshiba
internal call.
Asterisk - SETUP
Toshiba - CALL PROCESSING
Toshiba - CONNECT
Asterisk - CONNECT ACKNOWLEDGE
However when trying to dial 9 + number I received the following
Asterisk - SETUP
Toshiba - SETUP ACKNOWLEDGE
Looking at http://tools.ietf.org/html/rfc4497 I see the following
On receipt of a QSIG SETUP message containing no Sending complete
information element and a number in the Called party number
information element that the gateway cannot determine to be complete,
the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message, start
QSIG timer T302, and await further number digits.
Otherwise, the gateway SHALL wait for more digits
to arrive in QSIG INFORMATION messages.
Looking in the chan_dahdi.c code I see
case PRI_EVENT_SETUP_ACK:
chanpos = pri_find_principle(pri,
e-setup_ack.channel);
if (chanpos 0) {
ast_log(LOG_WARNING, Received
SETUP_ACKNOWLEDGE on unconfigured channel %d/%d span %d\n,
PRI_SPAN(e-setup_ack.channel), PRI_CHANNEL(e-setup_ack.channel),
pri-span);
} else {
chanpos =
pri_fixup_principle(pri, chanpos, e-setup_ack.call);
if (chanpos -1) {
ast_mutex_lock(pri-pvts[chanpos]-lock);
pri-pvts[chanpos]-setup_ack = 1;
/* Send any queued digits */
for (x = 0;x
strlen(pri-pvts[chanpos]-dialdest); x++) {
ast_debug(1,
Sending pending digit '%c'\n, pri-pvts[chanpos]-dialdest[x]);
pri_information(pri-pri, pri-pvts[chanpos]-call,
pri-pvts[chanpos]-dialdest[x]);
}
ast_mutex_unlock(pri-pvts[chanpos]-lock);
} else
ast_log(LOG_WARNING,
Unable to move channel %d!\n, e-setup_ack.channel);
}
break;
How do I get Asterisk to queue these digits so DAHDI can send them in
response to the SETUP ACKNOWLEDGE message. What should be happening is
Asterisk sends 9 via the SETUP message, waits for the SETUP
ACKNOWLEDGE, then send the 10 digits number via a INFORMATION message.
Ryan
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Re: [asterisk-users] Blind transfers security
Stanisław Pitucha s...@gradwell.net writes: I'm not sure that actually answers my question... If you have a A-B call and set accountcode for A on it, then B does a blind transfer, how do you set the correct accountcode then? (assuming B is a different customer and blind-transfers you to pstn) In modern versions of Asterisk, you can set other variables apart from accountcode in sip.conf. I'm fairly sure accountcode gets set to B in the blind transfer scenario you mention, but I'm not sure whether other variables do. Even if not, you can use some other variable for billing A-B, and accountcode for billing B-C. However, we simply do billing on our interconnect-Asterisks, where transfers aren't allowed. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfers security
On 15/09/09 3:12 AM, Stanisław Pitucha wrote: 2009/9/14 Olle E. Johanssono...@edvina.net: Make sure that each device has a TRANSFER_CONTEXT dialplan variable. What about a situation where sip devices register at a proxy in front of many asterisks and asterisks authorise all calls from that proxy? I.e. I don't have any devices that asterisk would know about. That way as far as asterisk is concerned, the call is a simple trunk call and the B side (in A-B call) doesn't trigger any TRANSFER_CONTEXT setting when doing a transfer. If your users are not connected to Asterisk and Asterisk just sees all calls as origination from your proxy, surely the place to sort this out would be the proxy. Can you not set a variable in the proxy before sending the call to Asterisk and use the sip header function to retrieve it once in Asterisk? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Aastra - Alert-Info : how to stop auto-answer on call second leg ?
Hi, When implementing click2dial feature, I can trigger an Aastra phone to auto-answer using statement like : SIPAddHeader(Alert-Info: info=alert-autoanswer); This is very convenient when trying to reach a distant party (ie through PSTN) The trouble is when 2 Aastra are calling each other over the LAN, this single statement is memorized somehow and both phones (caller and callee) auto-answer. Is there a way to cancel this auto-answer feature on the second leg of a call, either with a SIPRemoveHeader-like application or using something like (before dialing the second leg) : SIPAddHeader(Alert-Info: info=alert-noautoanswer); I've tried many things unsuccessfully such as: SIPAddHeader(Alert-Info: info=alert-community-1);(From an old doc) Best regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra - Alert-Info : how to stop auto-answer on call second leg ?
Add this line to your aastra.cfg file sip intercom allow barge in: 0 # don't barge in on existing calls Olivier wrote: Hi, When implementing click2dial feature, I can trigger an Aastra phone to auto-answer using statement like : SIPAddHeader(Alert-Info: info=alert-autoanswer); This is very convenient when trying to reach a distant party (ie through PSTN) The trouble is when 2 Aastra are calling each other over the LAN, this single statement is memorized somehow and both phones (caller and callee) auto-answer. Is there a way to cancel this auto-answer feature on the second leg of a call, either with a SIPRemoveHeader-like application or using something like (before dialing the second leg) : SIPAddHeader(Alert-Info: info=alert-noautoanswer); I've tried many things unsuccessfully such as: SIPAddHeader(Alert-Info: info=alert-community-1);(From an old doc) Best regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The o dial option
Hello, all. I see there is an o option for the Dial() command which reverts to the previous behavior of using the original callerid throughout the call - I suppose more specifically, using the callerid from leg 1 for leg 2 in B2BUA if I understand it correctly. That seems to be highly desirable behavior; I know we are seeing some problems with call history and call forwarding because of the default use of callerid. However I'm assuming it was changed to the current behavior for a good reason. Before we revert to the old behavior, I'd like to ask, why was it changed? What problems arose from the old behavior that provoked the change? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The o dial option
Hello, This changed years ago, and originally it was the 'p' dial option(for preserve CallerID). The reason we are told for the change was for calls being transferred within a company that originated on outside lines, so that you would know who the transfer was coming from. I didn't understand it either, but there it is. MATT--- On 9/14/09, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I see there is an o option for the Dial() command which reverts to the previous behavior of using the original callerid throughout the call - I suppose more specifically, using the callerid from leg 1 for leg 2 in B2BUA if I understand it correctly. That seems to be highly desirable behavior; I know we are seeing some problems with call history and call forwarding because of the default use of callerid. However I'm assuming it was changed to the current behavior for a good reason. Before we revert to the old behavior, I'd like to ask, why was it changed? What problems arose from the old behavior that provoked the change? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 for Asterisk
Thanks Ivan Stepaniuk. Thanks Vinicius for the the clear explanation. So we use this codec because it have good quality and many VOIP providers use it.(for interoperabilty) because Asterisk dont support this codec by default and we have to buy a lisence for it(per channel basis). Thanks Alot Adam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simple Time of Day Branching problem
Greetings folks, new to this, trying to get the syntax correct for a day of week routing. exten = 345,1,Answer() exten = 345,n,GotoIfTime(10:00-17:00|tuethusat|*|*?open,345,1) exten = 345,n,GotoIfTime(10:00-19:00|wedfri|*|*?open,345,1) exten = 345,n,Playback(afterhours) exten = 345,n,Hangup() I'll get an error stating incorrect day of week tuethursat, assuming none What is the correct syntax for this? We have longer hours on Wednesday and Fridays and we're closed Sunday/Monday Just trying to automate the time of day greeting etc. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dCAP Exam
Hi folks, Is there anywhere I can possibly get a model of the exam itself, maybe possible scenarios for the prac, etc? To people who have done the examany helpful hints ? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Time of Day Branching problem
It's easier to work with the closed hours then - use a goto just for Sunday/Monday PaulH James Hankins wrote: Greetings folks, new to this, trying to get the syntax correct for a day of week routing. exten = 345,1,Answer() exten = 345,n,GotoIfTime(10:00-17:00|tuethusat|*|*?open,345,1) exten = 345,n,GotoIfTime(10:00-19:00|wedfri|*|*?open,345,1) exten = 345,n,Playback(afterhours) exten = 345,n,Hangup() I'll get an error stating incorrect day of week tuethursat, assuming none What is the correct syntax for this? We have longer hours on Wednesday and Fridays and we're closed Sunday/Monday Just trying to automate the time of day greeting etc. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
