[asterisk-users] Remote host can't match request NOTIFY???
Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. They always come in pairs! I've seen a few other posts regarding these - some quite old - but no clear resolution or explanation. Could someone attempt to explain what it means and how I stop it? 10.0.0.2 btw is the IP address of the asterisk server. I'm running Asterisk 1.4.13. Cheers Al -- The way out is open! http://www.theopensourcerer.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
Re: [asterisk-users] Siemens Gigaset S685IP Review
Olivier wrote: Do we agree on the fact you can't change a S68 handset display name (S68 should be the model name of the handset included in a S685IP package) from a computer ? If my memory serves me right, you can change S685IP base station settings but not handset settings (display name, subscription to an other base station, ...), isn't it ? You can certainly give each handset a more sensible name if that's what you mean... From the web interface to the base station you can do this easily. My three handsets are now called Alan's, Helen's and Kitchen. You can, according to the manual subscribe each handset to up to 4 different base stations. The actual process of registering the handset requires manually pressing the Page button on the relevant base station, whilst the handset is in register me mode... Hope this helps. Al -- The way out is open! http://www.theopensourcerer.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
Re: [asterisk-users] Zaptel Compatibility
Mik Cheez wrote: Hmph...and it appears no kernel-smp-source exists. You should be able to compile going to a non-SMP kernel, but there must be a better solution. I can't believe this hasn't come up before. Sorry. You only need the kernel headers in reality I believe. Why not just mail RH and ask them for the headers? Al -- The way out is open! http://www.theopensourcerer.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
[asterisk-users] Newbie: How to remote test a call prolem in an Asterisk site?
I will be installing Asterisk in a few offices which I don't have any colleagues over there to help me. Let's suppose I installed Asterisk in such a site. I tested it to my satisfaction and I went back to my home office. One day, a customer called me to say that he had a problem calling out or something. Is there any way I could test out the problem myself remotely (apart from READING the message on the console when the customer tested) or I just have to believe what the customer tells me? Can anyone share their experience with me please? ___ -- 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] Was; Multiple Hunt Group to same extension advice Now: CallerID insert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Guy
Sent: 30 April 2008 18:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Multiple Hunt Group to same extension advice
Swapping out our old Mitel 3300 for Asterisk I need to come up with a
solution to advertise to the extension what hunt group the call was
for,
plus distinguishing those calls from calls that are sent straight to
the
extension.
OK I answered my own question. Use CALLERID NAME in the exten = except
that the Grandstream 1200 GXP I have on test will only disable the
CALLERID NAME not the NUMBER. Well it certainly does from extension to
extension, I currently can not hook it up to the PSTN
The Zoiper software phone does its job and displays both.
So I used exten = xxx,n,Set(CALLERID(name)=Technical:${CALLERID(name)})
Cracking
Cheers
Tim
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[asterisk-users] Queues v Multi Device dialing
Creating some 'hunt groups' Im not sure if to go for queues or just dial multi devices (one group is around 15 handsets) I don't need the queue features, although I am backgrounding thank you for holding every 30 seconds. Does one solution create more processor / networking overheads than another, or the fact I'm ringing the same amount of phones in both solutions mean they are equal? Cheers Tim This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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] Penalty based Cascading Queue - possible ?
Hi, folks. I have a call queue called 'support' with several members, for example : [support] member = Agent/65,5 member = Agent/78,5 member = Agent/74,1 member = Agent/62,1 With this configuration, I can configure an extension to send calls to agent 74 and 62 when they are logged in, and calls to Agent 65 and 78 when the first agents are busy, or not logged in. This works perfectly. I would like to configure a queue such that if Agents 74 and 62 do not answer, then the call is then presented to all of the four agents. This is described on voip-info as a cascading queue, and it's normally configured such that an extension to call the queue is made with a timeout, and if this fails, the call is presented to an alternative queue. This is far from ideal. I would like the call to be presented to the *same* queue, but to be able to specify a penalty that is associated with which queue members receive the call, e.g. exten = s,1,Answer exten = s,n,Queue(support|t|||10) -- penalty 1 gets the call this time .. exten = s,n,Queue(support) --- but somehow specify penalty 5 and below here Is this possible ? Many thanks Andy ___ -- 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] Newbie: How to remote test a call prolem in an Asterisk site?
On 1 May 2008, at 08:17, Lee, John (Sydney) wrote: I will be installing Asterisk in a few offices which I don't have any colleagues over there to help me. Let's suppose I installed Asterisk in such a site. I tested it to my satisfaction and I went back to my home office. One day, a customer called me to say that he had a problem calling out or something. Is there any way I could test out the problem myself remotely (apart from READING the message on the console when the customer tested) or I just have to believe what the customer tells me? Can anyone share their experience with me please? Send the logs to a syslog server. logger.conf [logfiles] syslog.local0 = debug, warning, error, notice, verbose ... then configure a syslog program on the asterisk box to send syslogs to your centralised syslog server that you use for clients you support. You will then be able to see the log messages generated on your own equipment, without needing access to the asterisk box. However, you will need to log into the asterisk box to make changes as per your customers' requirements ! Andy ___ -- 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] Newbie: How to remote test a call prolem in an Asterisk site?
On Thu, May 01, 2008 at 05:17:17PM +1000, Lee, John (Sydney) wrote: I will be installing Asterisk in a few offices which I don't have any colleagues over there to help me. Let's suppose I installed Asterisk in such a site. I tested it to my satisfaction and I went back to my home office. One day, a customer called me to say that he had a problem calling out or something. Is there any way I could test out the problem myself remotely (apart from READING the message on the console when the customer tested) or I just have to believe what the customer tells me? Can anyone share their experience with me please? If you have access to the console you can do many things. For instance, you can originate test calls. Sensible use of log messages and debug messages is also very handy for troubleshooting. e.g: most channel drivers provide a verbose lower-level debug (sip, iax2, pri). There are other things you can do, depending on the specific issue. Your question is quite generic. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] one way audio after call transfer
Do you mean the problem is solved using asterisk 1.4.18? Are you using the setting as mine? Below is my setting. One way audio is a result after A B connected. PSTN (A)--1200P-- Asterisk -- GXP2000 --blind transfer -- Extension B You can see that involve many parties in the blind transfer operation. I am not sure the problem is related to 1200P, Asterisk or GXP2000. That's why I seeking the solution from any person who touch the same problem before. asterisk version: asterisk 1.4.15 zaptel 1.4.7 asterisk addons 1.4.5 On Thu, May 1, 2008 at 4:49 AM, Duncan Turnbull [EMAIL PROTECTED] wrote: I had a similar issue in 1.2 after transfer and we were using SIP only but an upgrade cured it We are now on 1.4.18 still without issues Cheers Duncan ___ -- 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] ast_indicate_data: Unable to handle indication 3
Hi guys,
When I try to get ring tones when dialing out with the command
Dial(SIP/sipout/${PHONE},15,r), I get the error message indicated in the
subject. I've checked my indications.conf file using the sample file
provided with asterisk 1.4.10 (the version I'm using) and it's not better.
Any idea ?
Regards.
--
Cyril SCETBON
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[asterisk-users] TEST MAIL
sorry just a testmail to the list, becausemy last mail does not show up on the list. ___ -- 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] realtime queue callers
--- Atis Lezdins [EMAIL PROTECTED] wrote: So, the issue is http://bugs.digium.com/view.php?id=12556, feel free to comment about usage. I also posted backport to 1.4.19 at http://ftp.iq-labs.net/realtime_queue_callers-1.4/ but for this You will need to also apply backport for realtime store/destroy - also available at http://ftp.iq-labs.net/realtime_store_destroy-1.4/ Thank you for your contribution. I've posted a comment at http://bugs.digium.com/view.php?id=12556 and I invite other users with this particular feature request to post there too. If you solved the problem with another technique then please let us know. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9
Hi, Did you resolve the issue you were hitting ? aby azid wrote: hi, I hope anyone can tell me how to handle this kind of status message WARNING[13915]: chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9 [Apr 21 13:29:37] WARNING[13915]: channel.c:2390 ast_indicate_data: Unable to handle indication 9 for 'SIP/wellsip_kl-0833c1a8' [Apr 21 13:29:38] WARNING[13915]: chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9 [Apr 21 13:29:38] WARNING[13915]: channel.c:2390 ast_indicate_data: Unable to handle indication 9 for 'SIP/wellsip_kl-0833c1a8' I'm getting this message when call connected to SIP Proxy and to quintum gateway. In this case, I'm using Wellsip from Welltech. cheers, Aby Azid On Sun, Apr 20, 2008 at 11:54 PM, Eric Wieling [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Use the indications.conf.sample that comes with the Asterisk source. aby azid wrote: Thank you for replying, How would i know, whether i have the valid indicitions.conf ? On Sun, Apr 20, 2008 at 8:47 PM, Eric Wieling [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Make sure you have a valid /etc/asterisk/indications.conf aby azid wrote: Hi, this is my first ever post, would appreciate if anyone can explain it to me this status message: *[Apr 20 19:12:31] WARNING[759]: chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9 [Apr 20 19:12:31] WARNING[759]: channel.c:2390 ast_indicate_data: Unable to handle indication 9 for 'SIP/quintum_kl-0940c570' [Apr 20 19:12:32] WARNING[759]: chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9 [Apr 20 19:12:32] WARNING[759]: channel.c:2390 ast_indicate_data: Unable to handle indication 9 for 'SIP/quintum_kl-0940c570' *this happens when I sent call to my quintum gateway server, the status appears as soon as the call get connected. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 -- Cyril SCETBON ___ -- 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] Remote host can't match request NOTIFY???
Grey Man wrote: On Thu, May 1, 2008 at 7:54 AM, Alan Lord [EMAIL PROTECTED] wrote: Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. snip / Asterisk does not correctly match SIP NOTIFY transactions in at least some cases. Your problem may be related to http://bugs.digium.com/view.php?id=11848. Regards, Greyman. Thanks for that. Not sure I understand it all. I am not actually doing anything when these messages appear. They occur pretty much every minute or so. With or without any calls... Al -- The way out is open! http://www.theopensourcerer.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
Re: [asterisk-users] Remote host can't match request NOTIFY???
On Thu, May 1, 2008 at 1:38 PM, Alan Lord [EMAIL PROTECTED] wrote: Grey Man wrote: On Thu, May 1, 2008 at 7:54 AM, Alan Lord [EMAIL PROTECTED] wrote: Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. snip / Asterisk does not correctly match SIP NOTIFY transactions in at least some cases. Your problem may be related to http://bugs.digium.com/view.php?id=11848. Regards, Greyman. Thanks for that. Not sure I understand it all. I am not actually doing anything when these messages appear. They occur pretty much every minute or so. With or without any calls... There was a post week ago (I was having the same problem). For me it was caused by AudioCodes not understanding voicemail notifications. So, first You can enable SIP debug to see what packets are causing this, and if it's voicemail notifications, turn them off in sip.conf (mailbox= line). Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9
hi, yes, apparently the status message appeared due to the codec setting from the gateway. cheers, Aby Azid On Thu, May 1, 2008 at 5:34 PM, Cyril SCETBON [EMAIL PROTECTED] wrote: Hi, Did you resolve the issue you were hitting ? aby azid wrote: hi, I hope anyone can tell me how to handle this kind of status message WARNING[13915]: chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9 [Apr 21 13:29:37] WARNING[13915]: channel.c:2390 ast_indicate_data: Unable to handle indication 9 for 'SIP/wellsip_kl-0833c1a8' [Apr 21 13:29:38] WARNING[13915]: chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9 [Apr 21 13:29:38] WARNING[13915]: channel.c:2390 ast_indicate_data: Unable to handle indication 9 for 'SIP/wellsip_kl-0833c1a8' I'm getting this message when call connected to SIP Proxy and to quintum gateway. In this case, I'm using Wellsip from Welltech. cheers, Aby Azid On Sun, Apr 20, 2008 at 11:54 PM, Eric Wieling [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Use the indications.conf.sample that comes with the Asterisk source. aby azid wrote: Thank you for replying, How would i know, whether i have the valid indicitions.conf ? On Sun, Apr 20, 2008 at 8:47 PM, Eric Wieling [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Make sure you have a valid /etc/asterisk/indications.conf aby azid wrote: Hi, this is my first ever post, would appreciate if anyone can explain it to me this status message: *[Apr 20 19:12:31] WARNING[759]: chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9 [Apr 20 19:12:31] WARNING[759]: channel.c:2390 ast_indicate_data: Unable to handle indication 9 for 'SIP/quintum_kl-0940c570' [Apr 20 19:12:32] WARNING[759]: chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9 [Apr 20 19:12:32] WARNING[759]: channel.c:2390 ast_indicate_data: Unable to handle indication 9 for 'SIP/quintum_kl-0940c570' *this happens when I sent call to my quintum gateway server, the status appears as soon as the call get connected. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 -- Cyril SCETBON ___ -- 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] zap not coming online on fedora 8
On Thu, May 1, 2008 at 8:42 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Steve; My rc.local file as below, actually I have one digium card with 2 fxs and 2 fxo ports. Based on ur rc.local, I was not able which lines I have to add it (as I am using different card 2fxs+2fxo). Any reference can help? For example, what does it mean to use /sbin/modprobe qozap and why u used next to it /usr/sbin/wanrouter start and what that wanrouter. Any help? Please look for my current rc.local file. Bilal, qozap is the driver for the BRI card and wanrouter is for the Sangoma card. Your's will be a bit different since you are not using either. The one below should work for you I would think. touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Thanks, Steve Totaro ___ -- 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] zap not coming online on fedora 8
No problem, see comments inline. On Thu, May 1, 2008 at 9:14 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Steve; Really big thanks for the big efforts u gave me. I have here some topics, highly appreciate if u can advise: 1) What is fxotune -s and why to use the s argument? Google and the wiki are your friends. http://www.voip-info.org/wiki/view/Asterisk+fxotune Using Fxotune: You need to train fxotune once using fxotune -i, see more below about that. The training will create the file /etc/fxotune.conf . In order to apply the same tuning next time, you need to run: fxotune -s 2) Why we did not use modprobe zaptel? When you load wctdm, it will also load zaptel. 3) U r using sangoma, and I am using digium, the question is: zaptel can work with any telephony cards that support fxs and fxo modules? Like dialogic cards? The problem with zaptel or with the cards that they should be ready to work with us? The first part of your question is yes and no. With patches or additional drivers, zaptel works with other cards (such as Sangoma and various BRI cards). I think Dialogic may be supported now but I am not sure, maybe in ABE, again I am not sure. I am not sure if I understand the last part of your question. I think just about any card should work for you provided it has drivers for Asterisk. Regards Bilal --- Steve Totaro [EMAIL PROTECTED] wrote: On Thu, May 1, 2008 at 8:42 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Steve; My rc.local file as below, actually I have one digium card with 2 fxs and 2 fxo ports. Based on ur rc.local, I was not able which lines I have to add it (as I am using different card 2fxs+2fxo). Any reference can help? For example, what does it mean to use /sbin/modprobe qozap and why u used next to it /usr/sbin/wanrouter start and what that wanrouter. Any help? Please look for my current rc.local file. Bilal, qozap is the driver for the BRI card and wanrouter is for the Sangoma card. Your's will be a bit different since you are not using either. The one below should work for you I would think. touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Thanks, Steve Totaro Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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 on Xen or Dedicated
On 01/05/2008 00:27, George Pajari [EMAIL PROTECTED] wrote: On Wed, 2008-04-30 at 13:11 +0100, Dee Lowndes wrote: ...Question is do I still need to worry about timing and if so can this be resolved in a Xen enviroment?... We're an ITSP and use OpenVZ to offer customers Virtual Private Asterisk Servers (see www.vpas.ca) -- the same idea as Virtual Private Servers in the Linux world but with Asterisk added. Because of our network architecture, we chose to put the Digium cards in dedicated (i.e. not virtualised) servers acting as gateways to several OpenVZ servers so that the base environment (called VE0 in OpenVZ nomenclature) does nothing but load the ztdummy module. All the client VEs communicate with one or more SBCs or media gateways (i.e. servers with Digium Quad-PRI cards) using SIP or IAX. Each virtual environment has access to a pseudo timer so they can run meetme conferences etc. How does the pseudo timer compare to having a Digium card when handling large number of calls in a meetme conference? Also have you tried OpenVZ with a digium card does it allow direct access to it? Works very very well. We've migrated existing Asterisk configurations from dedicated servers to OpenVZ virtual servers for customers who cannot tell the difference. And a lot cleaner and more secure than trying to run multi-tenant configurations/dialplans within a single asterisk instance (which we still do for some customers for historical reasons). I quiet like the sound of that as it does get a bit messy all on one asterisk instance. Sorry but we've no experience running Asterisk on Xen -- we looked at Xen way back when were deciding on which way to go and chose OpenVZ because it was (at least for us) easier to get running, easier to support ztdummy, and more efficient (i.e. thinner) than Xen. One other question is how does multi cpu's scale is it better to have a highspeed dual core or a lower speed quad core? We use both and given the modest load you're proposing, it won't matter -- get the cheapest. Our benchmarks showed that we get more bang for the buck with X3210 Quad Core Xeons than the dual cores and so that is what we've standardised on for now but YMMV. Thanks for the pointers. Dee ___ -- 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] PRI hangup certain outgoing calls
Steve Totaro wrote: It's worth a shot, you should be running the latest 1.4.x code anyways right? Bugs can manifest themselves in different ways, or possibly the poster did not explain the issue accurately. I think the bug fits your problem more than any other explanation. Steve, the issue appears fixed after a upgrade and reboot, though I do not know if the remote party have changed their setup. Thanks for your help -- Alastair Battrick ___ -- 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] Background ring
3rd attempt.. get the right list... Hi All, When I hairpin calls out to some networks (eg international or mobiles), there can be a long delay until the PSTN starts sending audio ring tones back. Is there a way I can have asterisk play ringtones until the PSTN really answers?? I've looked at Playtone(), Background() Playback.. Playtone looks like it should do the job, but I still get a long silence. I think the PSTN may be sending an Accept signal back halting the Playtone(). Thanks, Adrian ___ -- 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] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)
Steve Totaro wrote: My question is does ANYONE do ANY testing on these releases? It would seem that this bug is so paramount to the purpose of the code that had anyone taken a MINUTE to TEST, it would have been discovered IMMEDIATELY. Not if you already had a zaptel udev rules script installed on the system that's used as the test machine. This was a regression do to recent Makefile changes. A test for this problem has now been added to our pre-release regression testing. Matthew Fredrickson sigh. Thanks, Steve Totaro On Wed, Apr 30, 2008 at 8:41 AM, Steve Totaro [EMAIL PROTECTED] wrote: Sean Bright to Asterisk show details 4:47 PM (15 hours ago) There is a bug in 'make install' in Zaptel 1.4.10 that causes the devices to not be installed correctly. You can either install 1.4.9 or wait for 1.4.11 to be released. On Wed, Apr 30, 2008 at 8:22 AM, Zoran Milenkovic, Datatek d.o.o. [EMAIL PROTECTED] wrote: Hi list! I have RHEL Server rel5 (Tikanga) 2.6.18-8.el5 #1 SMP Fri Jan 26 14:15:21 EST 2007 i686 i686 i386 GNU/Linux with installed digium packets 1. Asterisk 1.4.19 2. Zaptel 1.4.10 3. Libpri 1.4.3 My Digium hardware is [EMAIL PROTECTED] ~]# zaptel_hardware pci::04:00.0 wctdm+ e159:0001 Wildcard TDM400P REV I ...and I have 2 FXO and 2 FXS ports installed inside the TDM400P card The problem is the asterisk doesn't recognize the Zap channels at all. The error is No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) and there is the original output form Astersik console: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/zoran-09f1bf90, Zap/3|20) in new stack [Apr 30 13:17:48] WARNING[2845]: channel.c:3001 ast_request: No channel type registered for 'Zap' [Apr 30 13:17:48] WARNING[2845]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/zoran-09f1bf90, ) in new stack == Spawn extension (local, 12, 2) exited non-zero on 'SIP/zoran-09f1bf90' And everything was working quite fine when I was on asterisk 1.2.13, previously installed on this very same server, same Digium card etc. The configurations are totaly the same, also. What could be the resolution of this problem? ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- 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] How i know the version of my vpmadt032 firmware
Hi How can i know the version of my vpmadt032 firmware? Thanks Ruben ___ -- 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] Digium PRI card hi-Z for sniffing?
Does anyone know if the Digium PRI cards can be configured or modified to have a high-impedance input on the RX pair? I would be interested in this in order to build a bi-directional PRI audio sniffer using two E1/T1 ports per trunk to be monitored. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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
Re: [asterisk-users] How i know the version of my vpmadt032 firmware
Ruben, Ruben Zamora wrote: How can i know the version of my vpmadt032 firmware? Currently, it is only printed in the kernel log (view with the command dmesg) when the driver is loaded. Shaun ___ -- 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] Zaptel Compatibility
.build file is missing in the kernel-source package. Solutions is: Once you have the appropriate kernel sources installed you will need to configure them. Execute the following commands: cd /lib/modules/`uname -r`/build make mrproper Execute one of the following commands based on your hardware configuration (again, the exact file names may vary): cp -f configs/kernel-2.4.2-i586.config arch/i386/defconfig cp -f configs/kernel-2.4.2-i586-smp.config arch/i386/defconfig cp -f configs/kernel-2.4.2-i686-enterprise.config arch/i386/defconfig Verify that the kernel Makefile EXTRAVERSION information matches the version that you are running with respect to smp support. make oldconfig make dep Similar to 'make cloneconfig' in SuSE Linux. On Thu, May 1, 2008 at 3:06 AM, Alan Lord [EMAIL PROTECTED] wrote: Mik Cheez wrote: Hmph...and it appears no kernel-smp-source exists. You should be able to compile going to a non-SMP kernel, but there must be a better solution. I can't believe this hasn't come up before. Sorry. You only need the kernel headers in reality I believe. Why not just mail RH and ask them for the headers? Al -- The way out is open! http://www.theopensourcerer.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] Asterisk - CRM Integration
Michiel van Baak wrote: Have a look at Covide: http://sourceforge.net/projects/covide /shameless_plug Wow, what a disaster of an open source project. Install docs are impossible to use. Many, many inaccuracies. I Never could get it working. If you want acceptance, better make it easy to install. I don't care how well it works once installed, if I have to spend hours just figuring out HOW to get it installed, I won't waste my time. Covide might be a good project, but I KNOW Suger-CRM is because I set it up in about 15 minutes and SAW that it was... Bill ___ -- 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] How i know the version of my vpmadt032 firmware
Shaun And what it the last version of that firmware? Thanks Ruben Shaun Ruffell escribió: Ruben, Ruben Zamora wrote: How can i know the version of my vpmadt032 firmware? Currently, it is only printed in the kernel log (view with the command dmesg) when the driver is loaded. Shaun ___ -- 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] How i know the version of my vpmadt032 firmware
Ruben Zamora wrote: And what it the last version of that firmware? 107 is the current version installed by default. ___ -- 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] Sending caller name out PRI?
On Tue, 2008-04-29 at 23:49 -0500, Peter A Eisch wrote: I have a PRI connected to a traditional PBX using NI-2 and a typical config (further below). When I call from a SIP/IAX phone to an extension on the PBX, only the number makes it through. If I plug that same port on the PBX to a carrier the PBX presents both name and number. Hints or pokes to relevant chapters in documentation? My config is essentially like one found here: http://www.voip-info.org/files/nortel-asterisk-0.2.pdf The author makes no reference to CNID, so I'm assuming that he wasn't bothered by it not working. You could set pri debug span 1 in the Asterisk CLI (assuming that this is the first PRI span) and see if the name is actually being transmitted to the PBX. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] Zaptel Compatibility
On Wed, Apr 30, 2008 at 08:05:20PM -0400, Andreas van dem Helge wrote: On Wed, Apr 30, 2008 at 4:57 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Apr 30, 2008 at 09:21:37PM +0300, Tzafrir Cohen wrote: On Wed, Apr 30, 2008 at 02:00:57PM -0400, Andreas van dem Helge wrote: Is Zaptel 1.4.10 compatible with RHEL 3 (2.4.21-53.ELsmp)? Because I can compile 1.2.20.1 just fine but 1.4 says: echo You do not appear to have the sources for the 2.4.21-53.ELsmp kernel installed. You do not appear to have the sources for the 2.4.21-53.ELsmp kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/zaptel-1.4.10' make: *** [all] Error 2 Yes kernel-source is installed, there is no kernel-devel. I read one account where if I use non-SMP kernel it might work. But there's no fun it that. 1.2 works why not 1.4? Failing getting 1.4 to work can I use Zaptel 1.2 with Asterisk 1.4? I think not but just wanted to make sure. Zaptel will look as the kernel source for (in this specific order) 1. Whatever you explicitly set in KSRC (if you did) 2. /lib/modules/$KVERS/build (if you set KVERS explicitly) 3. /lib/modules/`uname -r`/build 4. /usr/src/linux-2.4 5. /usr/src/linux 'build' in (2) and (3) is normally a symlink to the path of the kernel. I forgot to mention that there's an additional test done: the source directory found (KSRC) has to have a file called .config in it . Which is the first of those directories that you actually have? To better debug this, edit the Makefile. Find the line with that error message and add the word '$(KSRC)' (without quotes) to it. This should help you see what the makefile thought is the kernel source tree. /lib/modules/2.4.21-53.ELsmp/build [EMAIL PROTECTED] [/]# ll /lib/modules/2.4.21-53.ELsmp/build lrwxrwxrwx1 root root 35 Apr 30 04:48 /lib/modules/2.4.21-53.ELsmp/build - ../../../usr/src/linux-2.4.21-53.EL/ There's something wrong with this system usr/src/linux-2.4.21-53.EL/.build is missing and I get errors trying to do 'make cloneconfig' .build ? What .build? I wrote .config above. (A nice example for whoever wants to demonstrate the damage from top posting) If that directory has no .config file, then it is not a configured kernel source directory. If indeed there isn't such a file, You need to figure out where you have a matching kernel tree for your kernel. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Zaptel Compatibility
On Thu, May 01, 2008 at 11:59:01AM -0400, Andreas van dem Helge wrote: .build file is missing in the kernel-source package. Solutions is: Once you have the appropriate kernel sources installed you will need to configure them. Execute the following commands: cd /lib/modules/`uname -r`/build make mrproper Which deletes the .config file. Great idea. Execute one of the following commands based on your hardware configuration (again, the exact file names may vary): cp -f configs/kernel-2.4.2-i586.config arch/i386/defconfig cp -f configs/kernel-2.4.2-i586-smp.config arch/i386/defconfig cp -f configs/kernel-2.4.2-i686-enterprise.config arch/i386/defconfig Which one, exactly? Verify that the kernel Makefile EXTRAVERSION information matches the version that you are running with respect to smp support. make oldconfig make dep Similar to 'make cloneconfig' in SuSE Linux. On Thu, May 1, 2008 at 3:06 AM, Alan Lord [EMAIL PROTECTED] wrote: Mik Cheez wrote: Hmph...and it appears no kernel-smp-source exists. You should be able to compile going to a non-SMP kernel, but there must be a better solution. I can't believe this hasn't come up before. Sorry. You only need the kernel headers in reality I believe. Why not just mail RH and ask them for the headers? Al -- The way out is open! http://www.theopensourcerer.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 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 - CRM Integration
On Thursday 01 May 2008 11:04:03 Bill Andersen wrote: Michiel van Baak wrote: Have a look at Covide: http://sourceforge.net/projects/covide /shameless_plug Wow, what a disaster of an open source project. Install docs are impossible to use. Many, many inaccuracies. I Never could get it working. If you want acceptance, better make it easy to install. I think you just missed the point of open source. Projects are almost always I made this to satisfy a need for myself, and it's open for others to examine and contribute. If you see a need for an easy installation process, then by all means, you should contribute that. I don't care how well it works once installed, if I have to spend hours just figuring out HOW to get it installed, I won't waste my time. Covide might be a good project, but I KNOW Suger-CRM is because I set it up in about 15 minutes and SAW that it was... Ease of setup does not always translate to best-of-breed. -- Tilghman ___ -- 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] Zaptel 1.4.10.1 Released
The Asterisk.org development team has announced the release of Zaptel version 1.4.10.1. This release is a bug fix release for a regression in which the Zaptel udev rules were not installed correctly, as well as a few minor fixes in the xpp drivers. This release is available as a tarball as well as a patch against the previous release. It is available for download from downloads.digium.com. Thank you for your support! ___ -- 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 - CRM Integration
Wow, what a disaster of an open source project. Install docs are impossible to use. Many, many inaccuracies. i think you just need someone set it up for you ... think of it as an air conditionning system, you can use it but can never install it on your own unless you're from the field. ___ -- 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 - CRM Integration
i think you just need someone set it up for you ... think of it as an air conditionning system, you can use it but can never install it on your own unless you're from the field. I installed Linux on my own. I installed Asterisk on my own. I installed Apache on my own. I installed MySQL on my own. I installed qmail on my own. Shall I go on? All from source... (OK, Linux was from CD) No. The problem is the docs are all wrong on Covide's project. The web site says one thing, the readme another. Neither are correct. COULD I figure out what's not updated? Yes. But my point was that if they want people to try their project, they need to make it easy to try. Or people will go elsewhere. Nuf said. Sorry. Very OT. Bill ___ -- 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 - CRM Integration
Tilghman Lesher I think you just missed the point of open source. Projects are almost always I made this to satisfy a need for myself, and it's open for others to examine and contribute. If you see a need for an easy installation process, then by all means, you should contribute that. Oh, I understand the open source point. It's just that open source has evolved. Now days, there are more than one open source solutions to your needs. If an open source project wants to become popular enough to lure open source developers to the table, the initial impression had better be good. Poorly documented install procedures by those that ARE already involved will do nothing to convince others to get involved with the current developers poor practices... Ease of setup does not always translate to best-of-breed. No argument here! ___ -- 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] Zaptel 1.4.10.1 Released
Does anybody know if this version fixes the soft lockup during ztcfg using a TE200B? http://bugs.digium.com/print_bug_page.php?bug_id=12468 -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Development Team [EMAIL PROTECTED] Sent: Thursday, May 01, 2008 1:07 PM Subject: [asterisk-users] Zaptel 1.4.10.1 Released The Asterisk.org development team has announced the release of Zaptel version 1.4.10.1. This release is a bug fix release for a regression in which the Zaptel udev rules were not installed correctly, as well as a few minor fixes in the xpp drivers. This release is available as a tarball as well as a patch against the previous release. It is available for download from downloads.digium.com. Thank you for your support! ___ -- 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 - CRM Integration
No. The problem is the docs are all wrong on Covide's project. The web site says one thing, the readme another. Neither are correct. well you may be correct but we must admit one thing, it takes a lot of dedication to start continue a real project ... and only for that every developper must get all of our respect. ___ -- 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] Zaptel 1.4.10.1 Released
err, that should of read TE220B From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Matt Watson [EMAIL PROTECTED] Sent: Thursday, May 01, 2008 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zaptel 1.4.10.1 Released Does anybody know if this version fixes the soft lockup during ztcfg using a TE200B? http://bugs.digium.com/print_bug_page.php?bug_id=12468 -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Development Team [EMAIL PROTECTED] Sent: Thursday, May 01, 2008 1:07 PM Subject: [asterisk-users] Zaptel 1.4.10.1 Released The Asterisk.org development team has announced the release of Zaptel version 1.4.10.1. This release is a bug fix release for a regression in which the Zaptel udev rules were not installed correctly, as well as a few minor fixes in the xpp drivers. This release is available as a tarball as well as a patch against the previous release. It is available for download from downloads.digium.com. Thank you for your support! ___ -- 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] http://www.asteriskdocs.org/html/apas02.html
If one of the authors is listening: http://www.asteriskdocs.org/html/apas02.html lists usereqphone 2 times. One of the entries should really be useragent. And the example for usereqphone is wrong. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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] Minimum upload speed for Asterisk?
I'm running an Asterisk box that's connected to the world via 5MB down/384kB up cable internet service. I've noticed that the sound quality for both IAX and SIP calls sometimes starts to suffer. IVR prompts and MOH frequently have slight pauses from the outside, but sound fine from inside calls. Is 384kB up too slow? Is there any guidance for the minimum upload speed for an Asterisk box? Frank ___ -- 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] Digium PRI card hi-Z for sniffing?
On May 1, 2008 11:39:52 am Tony Mountifield wrote: Does anyone know if the Digium PRI cards can be configured or modified to have a high-impedance input on the RX pair? I would be interested in this in order to build a bi-directional PRI audio sniffer using two E1/T1 ports per trunk to be monitored. I looked in to this exact thing several years ago, and while I do believe that the hardware is capable of it (I don't have a card in front of me, but I believe the T1 termination is done inside the QuadFALC), the driver does not have this capability. I don't think it would be impossible to add, but it would take some work, and please keep in mind that this was a number of years ago that I did look at this on the TE405P, so my memory may be a little hazy. -A. ___ -- 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] Minimum upload speed for Asterisk?
On Thu, May 1, 2008 at 3:19 PM, Frank Tarczynski [EMAIL PROTECTED] wrote: Is 384kB up too slow? Probably not. Is there any guidance for the minimum upload speed for an Asterisk box? I'm guessing this is for just a few calls at a time, correct? I'd guess that rather than these quality issues being caused by cramped bandwidth, they're actually being caused by latency issues. Have you ever checked the latency of the connection between your asterisk server and your SIP/IAX endpoint? If it's really high (say 300ms+) or if the latency is really erratic, you'll have quality issues. You didn't mention whether you are doing traffic shaping on your upstream connection, so I'll assume you're not. That would be something good to look into - with traffic shaping, you can prioritize your VoIP traffic over all other types of network traffic. -erik ___ -- 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] Providers
Good morning I would like to know which the best provider VOIP, that has quality and good prices for international calls. thank you. ___ -- 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] Zaptel 1.4.10.1 Released
Matt Watson wrote: Does anybody know if this version fixes the soft lockup during ztcfg using a TE200B? http://bugs.digium.com/print_bug_page.php?bug_id=12468 No, continue to use the stackcleanup branch. That is going to be merged in for the next major release (1.4.11). Matthew Fredrickson -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Development Team [EMAIL PROTECTED] Sent: Thursday, May 01, 2008 1:07 PM Subject: [asterisk-users] Zaptel 1.4.10.1 Released The Asterisk.org development team has announced the release of Zaptel version 1.4.10.1. This release is a bug fix release for a regression in which the Zaptel udev rules were not installed correctly, as well as a few minor fixes in the xpp drivers. This release is available as a tarball as well as a patch against the previous release. It is available for download from downloads.digium.com. Thank you for your support! ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- 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] Sending caller name out PRI?
On Thu, 1 May 2008, Jared Smith wrote:
You could set pri debug span 1 in the Asterisk CLI (assuming that this
is the first PRI span) and see if the name is actually being transmitted
to the PBX.
Thanks for your response.
Yes, with 'pri intense debug span 1' I do see the name in the setup
message. With it is the ANI as well as the DID in that specific.
Is there a way to delay (or resend) the name much like the carrier does?
This would then be closer to what the carrier does (as in how I need to
have a Wait(1) before using ${CALLERID(name)}). This assumes that it's a
timing issue I guess.
peter
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Re: [asterisk-users] Minimum upload speed for Asterisk?
You also need to check for Packet Loss on the Link Erik Anderson wrote: On Thu, May 1, 2008 at 3:19 PM, Frank Tarczynski [EMAIL PROTECTED] wrote: Is 384kB up too slow? Probably not. Is there any guidance for the minimum upload speed for an Asterisk box? I'm guessing this is for just a few calls at a time, correct? I'd guess that rather than these quality issues being caused by cramped bandwidth, they're actually being caused by latency issues. Have you ever checked the latency of the connection between your asterisk server and your SIP/IAX endpoint? If it's really high (say 300ms+) or if the latency is really erratic, you'll have quality issues. You didn't mention whether you are doing traffic shaping on your upstream connection, so I'll assume you're not. That would be something good to look into - with traffic shaping, you can prioritize your VoIP traffic over all other types of network traffic. -erik ___ -- 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] Sound Prompt 'per'
Anyone know where I can find an Alison recording of the word 'per'? Seems silly to buy the word 'per' from Digiums web site. And, I'd rather not open up audio editing software and get my 'per' prompt by editing it out of something else. Doug. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- 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] New generic sounds
We're about to do another batch of sounds, and I see by my word count that we have some extra time left over. So, suggestions will be entertained for additional prompts in English, Spanish, or French. The only rules are: 1) the prompts have to be generic to Asterisk. No Welcome to so-and-so's business unless the business is fake and the prompt is funny. 2) The prompt may not be profane. Our professional speakers do have a sense of humor, but there are some things they just will not say. I'll open it to the floor now, with the caveat that since Digium is paying for the recording session, it maintains final editorial approval over which sounds are selected. -- Tilghman ___ -- 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] Sound Prompt 'per'
On Thu, 1 May 2008, Douglas Garstang wrote: Anyone know where I can find an Alison recording of the word 'per'? Seems silly to buy the word 'per' from Digiums web site. And, I'd rather not open up audio editing software and get my 'per' prompt by editing it out of something else. How about Cepstral with the Allison voice font? It's only US$30. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New generic sounds
Tilghman Lesher wrote: We're about to do another batch of sounds, and I see by my word count that we have some extra time left over. So, suggestions will be entertained for additional prompts in English, Spanish, or French. The only rules are: 1) the prompts have to be generic to Asterisk. No Welcome to so-and-so's business unless the business is fake and the prompt is funny. 2) The prompt may not be profane. Our professional speakers do have a sense of humor, but there are some things they just will not say. I'll open it to the floor now, with the caveat that since Digium is paying for the recording session, it maintains final editorial approval over which sounds are selected. How about some prepaid balance-related ones that aren't calling-card-specific. Things like: Your balance is too low to connect this call. Please add additional funds to your account. Your account balance is... and one for the permissions set: ...from the account... (to go along with the Calls to the number you have dialed are not permitted) N. ___ -- 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] http://www.asteriskdocs.org/html/apas02.html
On Thu, 2008-05-01 at 22:10 +0200, Philipp Kempgen wrote: If one of the authors is listening: http://www.asteriskdocs.org/html/apas02.html lists usereqphone 2 times. One of the entries should really be useragent. And the example for usereqphone is wrong. Thanks! I'll get that fixed. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 - CRM Integration
Me too. I have a Java client which works with Asterisk and Salesforce... Justin -- From: Olivier [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk - CRM Integration To me, CRM-Asterisk integration has several meanings. It could refer to : - basic click2call feature from CRM contact or project panel, - journaling Asterisk incoming and outgoing calls inside CRM projects data, - programming and executing Conference calls defined inside CRM projects data - screen popup - etc... Which feature are you specifically looking for ? Do you plan to use it in a call center or casual business office ? Regards Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- 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] PCI ISDN as a PSTN gateway card
Is there any ISDN PCI cards that can be used with Asterisk as a PSTN gateway instead of using Diguim FXO cards?___ -- 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] New generic sounds
On Thu, 2008-05-01 at 18:25 -0400, SIP wrote: How about some prepaid balance-related ones that aren't calling-card-specific. Indeed, it would be nice to see the sounds supplied in the astcc package done by Allison. b. signature.asc Description: This is a digitally signed message part ___ -- 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 1.4.20-rc1 Now Available
The Asterisk development team has released Asterisk version 1.4.20-rc1. This release is a release candidate for the upcoming official release of 1.4.20. It contains a large number of bug fixes over the previous release, 1.4.19. We would like to encourage the community to assist us in testing before we release 1.4.20. The release candidate is available on the download site. http://downloads.digium.com/pub/telephony/asterisk Please provide release candidate testing feedback to the asterisk-dev mailing list, or the issue tracker, http://bugs.digium.com/. Thank you for your continued support of Asterisk! ___ -- 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] http://www.asteriskdocs.org/html/apas02.html
It's nice that the author is listening :) -- Jeffrey Thompson mailto:[EMAIL PROTECTED] POBOX 536, Suwanee, GA, 30024 770-234-8509 ___ -- 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] e164 Format Numbers
This is probably a very simple question, but I can't for the life of me work it out. I'm trying to use Asterisk as a PTSN gateway to OCS (and believe I have all the SIP issues sorted), but OCS wants to dial in e164 format (+613blahblah). Because Asterisk sees the + in the SIP URI, it doesn't want to match anything in my dial plan, not even the S extension in the nominated context. Am I missing something completely obvious? ___ -- 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] Stupid Timeout Question
I haven't done this for a while... yes, that is my excuse. What the heck is wrong with this? [general] autofallthrough=yes exten = s,n(prompt),NoOp() exten = s,n,Background(wish-to-continue) exten = s,n,Background(1-yes-2-no) exten = s,n,WaitExten(5) ; User entered nothing exten = t,1,Playback(yes-dear) exten = t,n,Goto(s,prompt) It never gets to the timeout extension when the user enters nothing. I tried it with autofallthrough set to no as well. No change. Asterisk 1.2. What am I missing? Doug. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- 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] Stupid Timeout Question
On Thu, 1 May 2008, Douglas Garstang wrote: What the heck is wrong with this? [general] autofallthrough=yes exten = s,n(prompt),NoOp() exten = s,n,Background(wish-to-continue) exten = s,n,Background(1-yes-2-no) exten = s,n,WaitExten(5) ; User entered nothing exten = t,1,Playback(yes-dear) exten = t,n,Goto(s,prompt) Waitexten() does not trigger a timeout when it expires, it continues inline. Add exten = s,n,verbose(Dooh!) exten = s,n,hangup() exten = 1,1,verbose(yes) exten = 1,n,hangup() exten = 2,1,verbose(no) exten = 2,n,hangup() after waitexten(). Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New generic sounds
The word Dialing... and Calling... As in Dialing 911, please wait... and as in Calling 911, please wait... Tilghman Lesher wrote: We're about to do another batch of sounds, and I see by my word count that we have some extra time left over. So, suggestions will be entertained for additional prompts in English, Spanish, or French. The only rules are: 1) the prompts have to be generic to Asterisk. No Welcome to so-and-so's business unless the business is fake and the prompt is funny. 2) The prompt may not be profane. Our professional speakers do have a sense of humor, but there are some things they just will not say. I'll open it to the floor now, with the caveat that since Digium is paying for the recording session, it maintains final editorial approval over which sounds are selected. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] e164 Format Numbers
Have you tried something like this?: exten = +12345,1,Noop(He died of ennui!) Rod Bacon wrote: This is probably a very simple question, but I can't for the life of me work it out. I'm trying to use Asterisk as a PTSN gateway to OCS (and believe I have all the SIP issues sorted), but OCS wants to dial in e164 format (+613blahblah). Because Asterisk sees the + in the SIP URI, it doesn't want to match anything in my dial plan, not even the S extension in the nominated context. Am I missing something completely obvious? ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] e164 Format Numbers
I did some dialplan work with numbers starting with + (outlook) and from memory things like exten = +X.,1,Answer Seemed to work fine... PaulH Melb, Australia On Fri, 2008-05-02 at 10:25 +1000, Rod Bacon wrote: This is probably a very simple question, but I can’t for the life of me work it out. I’m trying to use Asterisk as a PTSN gateway to OCS (and believe I have all the SIP issues sorted), but OCS wants to dial in e164 format (+613blahblah). Because Asterisk sees the “+” in the SIP URI, it doesn’t want to match anything in my dial plan, not even the S extension in the nominated context. Am I missing something completely obvious? ___ -- 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] Minimum upload speed for Asterisk?
On Thu, 01 May 2008 16:19:26 -0400, Frank Tarczynski wrote: I'm running an Asterisk box that's connected to the world via 5MB down/384kB up cable internet service. I've noticed that the sound quality for both IAX and SIP calls sometimes starts to suffer. IVR prompts and MOH frequently have slight pauses from the outside, but sound fine from inside calls. Is 384kB up too slow? No. But what you can do will depend upon codec selection, QoS traffic shaping in your router. Is there any guidance for the minimum upload speed for an Asterisk box? Rough guidance: G.711 needs approx 80 kbps per call leg G.729a needs approx 32 kbps per call leg See www.voip-info.org and search for codecs. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [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
Re: [asterisk-users] e164 Format Numbers
Unless you meant to match the literal +X., I think you meant to say: exten = _+X.,1,Answer (notice the leading underscore -- which indicates this is a pattern match) Paul Hales wrote: I did some dialplan work with numbers starting with + (outlook) and from memory things like exten = +X.,1,Answer Seemed to work fine... PaulH Melb, Australia On Fri, 2008-05-02 at 10:25 +1000, Rod Bacon wrote: This is probably a very simple question, but I can’t for the life of me work it out. I’m trying to use Asterisk as a PTSN gateway to OCS (and believe I have all the SIP issues sorted), but OCS wants to dial in e164 format (+613blahblah). Because Asterisk sees the “+” in the SIP URI, it doesn’t want to match anything in my dial plan, not even the S extension in the nominated context. Am I missing something completely obvious? ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] call pickup - Asterisk 1.4.19.1 -
Hello List, Does anyone here have call pickup (with *8 ) working ok on Asterisk version 1.4.19.1 ? Thanks in advice, -- Jose P. Espinal Slackware-Es.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
Re: [asterisk-users] e164 Format Numbers
agreed - typing stuff in emails while doing other things PaulH On Thu, 2008-05-01 at 21:17 -0500, Eric Wieling wrote: Unless you meant to match the literal +X., I think you meant to say: exten = _+X.,1,Answer (notice the leading underscore -- which indicates this is a pattern match) Paul Hales wrote: I did some dialplan work with numbers starting with + (outlook) and from memory things like exten = +X.,1,Answer Seemed to work fine... PaulH Melb, Australia On Fri, 2008-05-02 at 10:25 +1000, Rod Bacon wrote: This is probably a very simple question, but I can’t for the life of me work it out. I’m trying to use Asterisk as a PTSN gateway to OCS (and believe I have all the SIP issues sorted), but OCS wants to dial in e164 format (+613blahblah). Because Asterisk sees the “+” in the SIP URI, it doesn’t want to match anything in my dial plan, not even the S extension in the nominated context. Am I missing something completely obvious? ___ -- 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
Re: [asterisk-users] Digium PRI card hi-Z for sniffing?
Hi Tony, http://www.voicetronix.com.au/openpri.htm Never tested, though. We used the analogue boards for monitoring, so far. Jorge Tony Mountifield wrote: Does anyone know if the Digium PRI cards can be configured or modified to have a high-impedance input on the RX pair? I would be interested in this in order to build a bi-directional PRI audio sniffer using two E1/T1 ports per trunk to be monitored. Cheers Tony ___ -- 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] Stupid Timeout Question
Hi, It may have to do with the version of Asterisk. I have (basically) the same coding on an Asterisk V1.4.18 box, and a V1.6 SVN test box - in both boxes the Asterisk does execute the = t,1,Playback(connection-timed-out) when nothing is entered. The only differences I can see between your coding and mine, is that a) I simply use the default timeout (i.e., WaitExten() ); but don't see why this matters, and b) I use the m option in the Background command, since I have a one-key extension. You could try using, exten = s,n,Set(TIMEOUT(absolute)=5) before the Background command, and see if this works. Gerald H. ___ -- 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 - get Caller String(as per key action)
Dear Sir, I have a one query for Asterisk, I want to make a dial plan to the conference in Caller, Asterisk and my staff, and my staff will also transfer call to return PBX to IVR. and when caller make press key as a date of birth then ivr can make calculate sum in one digite (Example :- 12/12/2000 total is 8 ). So what I have to do in my dial-plan. My Asterisk System is transfer call to PSTN line. With Regards, Hiren Mistry. ___ -- 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
