Re: [asterisk-users] Visual Dial Plan
Hi All, Apologies for this, migrated the site and forgot to change a path. Site's back up now. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Sunday, July 27, 2008 8:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Visual Dial Plan randulo wrote: On Sun, Jul 27, 2008 at 12:19 PM, Matt Watson [EMAIL PROTECTED] wrote: I've seen it before infact there is a website setup where people can post stuff made with it... kind of super nerdy! http://www.ratemydialplan.com Cannot find lib path too nerdy! Agreed - no lib, no site! PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P FXO not seeing ringing after software update
A customer has an Asterisk box with two TDM400P cards, running [EMAIL PROTECTED] 2.8, which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok for a while, with only some small issues. They have FXO ports going to analogue POTS (UK standard) lines, and SIP phones for extensions. I just tried updating Zaptel to 1.2.26 and Asterisk to 1.2.30.1, and although the update appeared to go ok, the system would no longer detect incoming calls. It appeared not to see the ringing signal. Outgoing calls over the PSTN lines still worked fine. Does anyone know if there have been changes in the 1.2 series that affects ring detection on the TDM400P FXO ports? Any critical settings in zaptel.conf or zapata.conf? For now, I've had to revert to the original versions, in which ringing once again works fine. 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Line 0005 cannot be answered?
I have a Digium Appliance AA50 configure with 8 lines and two dial plans. Each dial plan takes care of a particular location. In Dialplan2 we have 4 lines. xxx xxx 0333 xxx xxx 0005 xxx xxx 0006 xxx xxx 0007 When a call gets to line 0005 you pick up the phone but the call does not get connected, you can still hear the phone ringing in a noisy weird way on the handset - the caller's phone never stops ringing. Is this a problem related to the way the line was taken out of the punch panel or does I have to do anything with configuration? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to test tftp for phones provisioning
Olivier wrote: tftp is the client, do you have it installed ?... example: # tftp hostname tftp get /srv/tftp/foo.txt tftp ^D # cat foo.txt ... That was exactly what I was after : I installed tftp on my Ubuntu system and checked Debian tftp server Things to check: is /srv/tftp the tftp directory or is it the os filesystem directory where the tftp root resides ? Also, the tftp daemon in CentOS is started by xinetd and can be invoked with extra -v flags so as to increase logging verbosity. Check your dist. This may help... Yes, that's the next step. I could see a tftp service is running ok on my server and I need to increase its logs to pinpoint root cause. Olivier, also check that you don't have any firewalls in the way, especially if there is nothing in the logs. Turn them off for testing, on both server and CLIENT. I found out about the client f/w blocking tftp the hard way! :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] custom configuration with appliance aa50.
I have just received a list of requests from one of our customers and I really do not have the time or knowledge to work on it. I will truly appreciate it if someone could help me outside of the mailing list. Please contact me if interested. Digium Appliance AA50/GrandStream GXP2000. Here is the list: 1. We need to be able to program the buttons along the right side of the phone so we can see who is on the phone. 2. How to implement phone paging/intercom. 3. If the front desk transfers a call to Jerry's phone but he isn't there how can I grab that call from my phone so it doesn't go to his voicemail? 4. If we transfer a call can we send it directly to voice mail or does it have to ring on the person's phone first? 5. How do we set after hour message and holiday messages? 6. Can we set a different ring tones for incoming calls and ext. to ext calls? 7. Is there an intercom on the phone for ext to ext. calls or do we have to pick up the handset to hear the person calling you? 8. When a call is transferred to another extension provide a recording that the caller can hear: Leave a message or press zero to return to the operator. If ZERO is entered can we go back to the extension the call came from? Instead of dropping it to the pool again. Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Line 0005 cannot be answered?
Have you tried to contact Digium Support about this question? They are very good at answering AA50 configuration questions. Regards, Doug Bailey - Fidel Garcia [EMAIL PROTECTED] wrote: I have a Digium Appliance AA50 configure with 8 lines and two dial plans. Each dial plan takes care of a particular location. In Dialplan2 we have 4 lines. xxx xxx 0333 xxx xxx 0005 xxx xxx 0006 xxx xxx 0007 When a call gets to line 0005 you pick up the phone but the call does not get connected, you can still hear the phone ringing in a noisy weird way on the handset - the caller’s phone never stops ringing. Is this a problem related to the way the line was taken out of the punch panel or does I have to do anything with configuration? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 06:33:35PM -0400, Eric ManxPower Wieling wrote: The something is generated by Asterisk at the time the call is created. You should never add it, since you don't control that call instance info. In fact, you should almost never care about the call instance string. The -1 means first instance of a call on this channel, a -2 would be seen in you answer a 2nd call for call waiting. Ah. Got it. Thanks. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callcentric Issues
Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get handle_request_invite: Failed to authenticate user sip:PSTNnumber This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf entry for callcentric. [callcentric-id] type=friend context=incoming host=callcentric.com dtmfmode=rfc2833 fromuser=my_callcentric_user fromdomain=callcentric.com secret=my_callcentric_pw insecure=port,invite qualify=no srvlookup=yes Those of you that use callcentric have you experienced this issue and if so how did you solve it? Thanks, Igor H. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callcentric Issues
emist wrote: Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get handle_request_invite: Failed to authenticate user sip:PSTNnumber This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf entry for callcentric. [callcentric-id] type=friend context=incoming host=callcentric.com dtmfmode=rfc2833 fromuser=my_callcentric_user fromdomain=callcentric.com secret=my_callcentric_pw insecure=port,invite qualify=no srvlookup=yes Those of you that use callcentric have you experienced this issue and if so how did you solve it? I had a similar problem. Inbound calls would complete about 50% of the time. After some digging I noticed the calls were coming from an IP that did not resolve to callcentric.com. I solved it by setting allowguest=yes and context=guests under [general] in sip.conf and adding my callcentric number to the guests context in extensions.conf. They actually show this in their support documentation for asterisk. It's not ideal, but the odds of someone other than callcentric placing a call to my asterisk box and calling a 1777xxx phone number are pretty slim. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to find out RTP UDP port of active calls
Hello list, I want to use Asterisk as a PBX connected to a public SIP service provider as uplink. The environment where I want to deploy the solution makes it necessary to request (IP guaranteed quality of service) resources per active call. This is why I am looking for a way to interface a resource management software (not yet developed) with Asterisk so that 1. the software is informed whenever an external call is about to be established or ends (where polling Asterisk is an acceptable alternative if there is no notification mechanism) 2. the software knows the RTP traffic's negotiated UDP port in order to distinguish (every single) voice streams from other traffic. Is there an interface in Asterisk that fulfils these two requirements? To satisfy (1.) I found the event mechanism in the Asterisk Manager Interface. However as far as I understood the Manager Interface only gives me SIP peers' SIP UDP port numbers but no RTP UDP port numbers. (Right?) If there is no such interface, my idea was to intercept the SIP INVITE, OK and BYE messages with libpcap, parse the SDP payload and retrieve the required information that way, but I hope there is a more adequate solution. Thanks for any hints and comments Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Common Inter-Queues Leastrecent Strategy
Hi list: Is there any way, to set a common inter-queues leastrecent Strategy, i'm searching a Behaviour like this: 2 Queues Q1 and Q2 2 Agentes A1 and A2 Both agents are in both queues. First Call in the system is for Q1 and is answer by A1 The next call in the system is for Q2, both agents are free, the system deliver the call to A1... but we want that the call be answer by the A2. Any idea??? Thanks. -- Alvaro I. Parres Peredo Director de IT Grupo Xmarts SA de CV Tel: +52 (33) 35 63 6261 Ext. 112 01 800 087 2260 Cel: +52 (33) 33 68 1087 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP sprials and 482 Loop Detected
Hi guys, I know this problem has just been fixed in trunk (http://bugs.digium.com/view.php?id=7403), but I'm asking for a workaround for previous versions of Asterisk, as we can't run off of Trunk (e.g. we have to run with Asterisk from Ubuntu 8.10). Basically, I have a situation where I have Openser and Asterisk running on the same box. I get a SIP call in (from a remote SIP trunk) to Asterisk. Asterisk then does a Dial() and sends the INVITE over to Openser, which rewrites the RURI and then sends the call back to Asterisk. With this, as the callID and To: fields are the same Asterisk gets confused and issues a 486 Loop detected error. Of course, the RURI is different but Asterisk doesn't see this. And yes, this may seem convoluted. Normally these components run on different machines, but I am testing a use case where we have them all on the same machine. When they're running on different machines (e.g. Asterisk on box A gets the request, passes it to Openser on box A, then Openser sends it to Asterisk on box B) this works without a problem. What I'm wondering is if there is some harmless way the Openser Proxy can mangle the INVITE back to the Asterisk box so that it avoids this Loop detected error we see here. Most likely any of this will violate the SIP spec, but it is a temporary solution so as far as this goes I'd just like something that works for now until that patch comes to Ubuntu. Any ideas? I have a SIP debug dump from the Asterisk box below. Note that I have Openser running on port 5060 and Asterisk running on port 5080. They both run on lab1-int-012 (10.1.14.12), which is 1 to 1 NATted through to 74.229.XXX.XXX. 64.85.162.136 is the IP of the SIP trunk. -- Connected to Asterisk 1.4.17~dfsg-2ubuntu1 currently running on lab1-int-012 (pid = 7543) Verbosity was 0 and is now 3 lab1-int-012*CLI --- SIP read from 64.85.162.136:5060 --- INVITE sip:[EMAIL PROTECTED]:5080 SIP/2.0 Via: SIP/2.0/UDP 64.85.162.136:5060;branch=z9hG4bK332bdcbc;rport From: 919932 sip:[EMAIL PROTECTED];tag=as36c86fee To: sip:[EMAIL PROTECTED]:5080 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: CWU SIP-GW Max-Forwards: 70 Date: Mon, 28 Jul 2008 14:03:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 385 v=0 o=root 4133 4133 IN IP4 208.44.220.234 s=session c=IN IP4 208.44.220.234 t=0 0 m=audio 14852 RTP/AVP 0 8 3 97 18 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv - --- (14 headers 18 lines) --- Sending to 64.85.162.136 : 5060 (NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found no matching peer or user for '64.85.162.136:5060' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 111 Found RTP audio format 101 Peer audio RTP is at port 208.44.220.234:14852 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format iLBC for ID 97 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 111 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0xd0e (gsm|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 208.44.220.234:14852 Looking for 1 in public (domain 74.229.XXX.XXX) list_route: hop: sip:[EMAIL PROTECTED] --- Transmitting (NAT) to 64.85.162.136:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.85.162.136:5060;branch=z9hG4bK332bdcbc;received=64.85.162.136;rport=5060 From: 919932 sip:[EMAIL PROTECTED];tag=as36c86fee To: sip:[EMAIL PROTECTED]:5080 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED]:5080 Content-Length: 0 -- Executing [EMAIL PROTECTED]:1] Answer(SIP/64.85.162.136-007414e0, ) in new stack Audio is at 74.229.XXX.XXX port 12660 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Reliably Transmitting (NAT) to 64.85.162.136:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 64.85.162.136:5060;branch=z9hG4bK332bdcbc;received=64.85.162.136;rport=5060 From: 919932 sip:[EMAIL PROTECTED];tag=as36c86fee To: sip:[EMAIL PROTECTED]:5080;tag=as47267a1a Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported:
Re: [asterisk-users] TDM400P FXO not seeing ringing after software update
At 01:58 AM 7/28/2008, you wrote: Does anyone know if there have been changes in the 1.2 series that affects ring detection on the TDM400P FXO ports? Any critical settings in zaptel.conf or zapata.conf? Some of us know there were changes because we've experienced the same failure. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] announcement server using asterisk
Quote Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Grüße, Philipp Kempgen Yes - There is also a lot of bogus, incorrect, crap. His question was fair, on-topic, politely asked and as such hardly deserves to be made fun off Dean Collins wrote: Lol crackup. Having said that here is some help. Don't even think of using a laptop that's just dumb. Next - check out www.voip-info.org you'll find what you need there. Regards, Dean Collins +1-212-203-4357 (Direct) +61-2-9016-5642 (Sydney in-dial) http://www.Cognation.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Saturday, 26 July 2008 9:57 AM To: Asterisk Users Subject: Re: [asterisk-users] announcement server using asterisk ballamudi madhulika schrieb: Can I use Asterisk as an announcement server. We want to build announcement server with ISDN PRI card terminating on our server and announcement being fed on the incoming calls. Yes. Also is there any ISDN card available for Laptop. Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Grüße, Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP sprials and 482 Loop Detected
Robby Dermody wrote: Hi guys, I know this problem has just been fixed in trunk (http://bugs.digium.com/view.php?id=7403), but I'm asking for a workaround for previous versions of Asterisk, as we can't run off of Trunk (e.g. we have to run with Asterisk from Ubuntu 8.10). Basically, I have a situation where I have Openser and Asterisk running on the same box. I get a SIP call in (from a remote SIP trunk) to Asterisk. Asterisk then does a Dial() and sends the INVITE over to Openser, which rewrites the RURI and then sends the call back to Asterisk. With this, as the callID and To: fields are the same Asterisk gets confused and issues a 486 Loop detected error. Of course, the RURI is different but Asterisk doesn't see this. snip Thanks, Robby Sorry if it was unclear, but that issue has been fixed in both Asterisk 1.4 and in trunk. There hasn't been a 1.4 release made yet which has the fix, but it will be in the upcoming 1.4.22 release. If you are using a subversion checkout of version 1.4, then you can update to any revision after 132790. Hopefully you'll see that the spirals work correctly. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO not seeing ringing after software update
On Mon, Jul 28, 2008 at 10:10:49AM -0700, Ira wrote: At 01:58 AM 7/28/2008, you wrote: Does anyone know if there have been changes in the 1.2 series that affects ring detection on the TDM400P FXO ports? Any critical settings in zaptel.conf or zapata.conf? Some of us know there were changes because we've experienced the same failure. Some of you filed a bug report? Do you happen to have a newer version than 1.2.5 where it worked well? -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] announcement server using asterisk
I really don't see any issue with running it on a laptop, you even have a built in UPS. That is of course if you need to be up with some guarantee. I have first gen Pentium based Dell Latitudes with several years of uptime. Anyways, with two laptops configured identically and this http://www.voipsupply.com/product_info.php?products_id=4069osCsid=3ea19c2324c85f2923deb8dfd25c2cf4 you can be pretty safe. Thanks, Steve Totaro On Mon, Jul 28, 2008 at 1:16 PM, Al Baker [EMAIL PROTECTED] wrote: Quote Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Grüße, Philipp Kempgen Yes - There is also a lot of bogus, incorrect, crap. His question was fair, on-topic, politely asked and as such hardly deserves to be made fun off Dean Collins wrote: Lol crackup. Having said that here is some help. Don't even think of using a laptop that's just dumb. Next - check out www.voip-info.org you'll find what you need there. Regards, Dean Collins +1-212-203-4357 (Direct) +61-2-9016-5642 (Sydney in-dial) http://www.Cognation.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Saturday, 26 July 2008 9:57 AM To: Asterisk Users Subject: Re: [asterisk-users] announcement server using asterisk ballamudi madhulika schrieb: Can I use Asterisk as an announcement server. We want to build announcement server with ISDN PRI card terminating on our server and announcement being fed on the incoming calls. Yes. Also is there any ISDN card available for Laptop. Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Grüße, Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO not seeing ringing after software update
Tony Mountifield wrote: A customer has an Asterisk box with two TDM400P cards, running [EMAIL PROTECTED] 2.8, which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok for a while, with only some small issues. They have FXO ports going to analogue POTS (UK standard) lines, and SIP phones for extensions. I just tried updating Zaptel to 1.2.26 and Asterisk to 1.2.30.1, and although the update appeared to go ok, the system would no longer detect incoming calls. It appeared not to see the ringing signal. Outgoing calls over the PSTN lines still worked fine. Does anyone know if there have been changes in the 1.2 series that affects ring detection on the TDM400P FXO ports? Any critical settings in zaptel.conf or zapata.conf? For now, I've had to revert to the original versions, in which ringing once again works fine. Cheers Tony YES! I have a 4xFXO board in Singapore. Having tried many various settings, etc., I had assumed it was the I18N of the ringtones and signalling. Does anyone know at which rev the issue appeared? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] simultaneous dial macro
Hi, Would just like to know if it's possible to be able to call a macro at the same time. i use a macro to dial local extension to another extension. exten = 100,Macro(dial-ext|SIP/100) exten = 101,Macro(dial-ext|SIP/101) but now i would like to use it on a simple ringgroup where it will ring all extensions e.g. exten = s,Dial(SIP/100SIP/101) how can i make use of my dial-ext macro instead of the simple Dial(SIP SIP SIP) thank you regards, ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Slow Playback of Recorded Files
OS: CentOS 5.2 Asterisk: 1.4 I use NeoSpeech to do TTS recording. When I play back the files in Asterisk, the playback seems to be at about half-speed or less. However, when I play them through totem, xmms or other audio applications they sound fine. I've tried recording in 8-bit Mu-law PCM and OGG. Other formats I have tried (16-bit linear PCM, 8-bit alaw, 16-bit linear PCM wave, 8-bit unsigned linear wave, etc) haven't played well at all (either not playing or producing static). Is there a generally known cause for slow audio playback? Thanks, -- Deric Page [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] announcement server using asterisk
You could pick up a couple of these cute little guys http://www.surpluscomputers.com/store/Main.aspx?p=ItemDetailitem=com10791 Make them identical and then use the single Redfone PRI box http://www.voipsupply.com/product_info.php?products_id=4069osCsid=3ea19c2324c85f2923deb8dfd25c2cf4 That is redundancy on the cheap! Thanks, Steve Totaro On Mon, Jul 28, 2008 at 1:50 PM, Steve Totaro [EMAIL PROTECTED] wrote: I really don't see any issue with running it on a laptop, you even have a built in UPS. That is of course if you need to be up with some guarantee. I have first gen Pentium based Dell Latitudes with several years of uptime. Anyways, with two laptops configured identically and this http://www.voipsupply.com/product_info.php?products_id=4069osCsid=3ea19c2324c85f2923deb8dfd25c2cf4 you can be pretty safe. Thanks, Steve Totaro On Mon, Jul 28, 2008 at 1:16 PM, Al Baker [EMAIL PROTECTED] wrote: Quote Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Grüße, Philipp Kempgen Yes - There is also a lot of bogus, incorrect, crap. His question was fair, on-topic, politely asked and as such hardly deserves to be made fun off Dean Collins wrote: Lol crackup. Having said that here is some help. Don't even think of using a laptop that's just dumb. Next - check out www.voip-info.org you'll find what you need there. Regards, Dean Collins +1-212-203-4357 (Direct) +61-2-9016-5642 (Sydney in-dial) http://www.Cognation.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Saturday, 26 July 2008 9:57 AM To: Asterisk Users Subject: Re: [asterisk-users] announcement server using asterisk ballamudi madhulika schrieb: Can I use Asterisk as an announcement server. We want to build announcement server with ISDN PRI card terminating on our server and announcement being fed on the incoming calls. Yes. Also is there any ISDN card available for Laptop. Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Grüße, Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO not seeing ringing after software update
Drew Gibson wrote: Tony Mountifield wrote: A customer has an Asterisk box with two TDM400P cards, running [EMAIL PROTECTED] 2.8, which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok for a while, with only some small issues. They have FXO ports going to analogue POTS (UK standard) lines, and SIP phones for extensions. I just tried updating Zaptel to 1.2.26 and Asterisk to 1.2.30.1, and although the update appeared to go ok, the system would no longer detect incoming calls. It appeared not to see the ringing signal. Outgoing calls over the PSTN lines still worked fine. Does anyone know if there have been changes in the 1.2 series that affects ring detection on the TDM400P FXO ports? Any critical settings in zaptel.conf or zapata.conf? For now, I've had to revert to the original versions, in which ringing once again works fine. Cheers Tony YES! I have a 4xFXO board in Singapore. Having tried many various settings, etc., I had assumed it was the I18N of the ringtones and signalling. Does anyone know at which rev the issue appeared? Tzafrir, Tony tried Zaptel 1.2.26, and I was at 1.2.25, both were failing to pickup. I just downgraded to 1.2.24 and the system is now picking up incoming calls. Asterisk remained at 1.2.28. From the Changelog file I found a possibly relevant change... 2008-04-04 04:29 + [r4126-4132] sruffell [EMAIL PROTECTED]: ... involved wctdm.c and fxo_modes.h. Merges with 1.4 code? Any use? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callcentric Issues
Hey Dave, Thanks for the help. Thats about the only thing I didn't think to try. It seems to have resolved the problem. Regards, Igor H. Dave Fullerton wrote: emist wrote: Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get handle_request_invite: Failed to authenticate user sip:PSTNnumber This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf entry for callcentric. [callcentric-id] type=friend context=incoming host=callcentric.com dtmfmode=rfc2833 fromuser=my_callcentric_user fromdomain=callcentric.com secret=my_callcentric_pw insecure=port,invite qualify=no srvlookup=yes Those of you that use callcentric have you experienced this issue and if so how did you solve it? I had a similar problem. Inbound calls would complete about 50% of the time. After some digging I noticed the calls were coming from an IP that did not resolve to callcentric.com. I solved it by setting allowguest=yes and context=guests under [general] in sip.conf and adding my callcentric number to the guests context in extensions.conf. They actually show this in their support documentation for asterisk. It's not ideal, but the odds of someone other than callcentric placing a call to my asterisk box and calling a 1777xxx phone number are pretty slim. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO not seeing ringing after software update
In article [EMAIL PROTECTED], Drew Gibson [EMAIL PROTECTED] wrote: Tony Mountifield wrote: A customer has an Asterisk box with two TDM400P cards, running [EMAIL PROTECTED] 2.8, which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok for a while, with only some small issues. They have FXO ports going to analogue POTS (UK standard) lines, and SIP phones for extensions. I just tried updating Zaptel to 1.2.26 and Asterisk to 1.2.30.1, and although the update appeared to go ok, the system would no longer detect incoming calls. It appeared not to see the ringing signal. Outgoing calls over the PSTN lines still worked fine. Does anyone know if there have been changes in the 1.2 series that affects ring detection on the TDM400P FXO ports? Any critical settings in zaptel.conf or zapata.conf? For now, I've had to revert to the original versions, in which ringing once again works fine. YES! I have a 4xFXO board in Singapore. Having tried many various settings, etc., I had assumed it was the I18N of the ringtones and signalling. Does anyone know at which rev the issue appeared? I've looked through all the revisions in svn.digium.com, comparing the tag revisions with the history of wctdm.c in zaptel/branches/1.2 My guess is that 1.2.24 will work, which was at revision 3842 of the tree (rev 3741 of wctdm.c). The next change to wctdm.c (rev 4126) looks innocuous enough, but the follwing two (revs 4128 and 4132) look likely culprits, from looking at the areas of code that they affect. I can't test the affected system until tomorrow, but installing 1.2.24 is the very first thing that I will try. After that, it is narrowing down which specific change causes the problem, but that will be more tricky to schedule, as this is a production system. 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO not seeing ringing after software update
Tony Mountifield wrote: My guess is that 1.2.24 will work, which was at revision 3842 of the tree (rev 3741 of wctdm.c). The next change to wctdm.c (rev 4126) looks innocuous enough, but the follwing two (revs 4128 and 4132) look likely culprits, from looking at the areas of code that they affect. It is very likely 4128, based on the code it affects and the behavior that is being reported. Please let us know as soon as you can (anyone who has this problem), if reverting r4128 from current Zaptel branch 1.2 SVN solves the problem. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] simultaneous dial macro
you can try to place your macro extensions into single dialgroup using
DIALGROUP() function and then reference that dialgroup in dial aplication,
eg.
Set(DIALGROUP(test,add)=Local/100)
Set(DIALGROUP(test,add)=Local/101)
Dial(${DIALGROUP(test)})
ronald ramos wrote:
Hi,
Would just like to know if it's possible to be able to call a macro at the
same time.
i use a macro to dial local extension to another extension.
exten = 100,Macro(dial-ext|SIP/100)
exten = 101,Macro(dial-ext|SIP/101)
but now i would like to use it on a simple ringgroup where it will ring all
extensions
e.g. exten = s,Dial(SIP/100SIP/101)
how can i make use of my dial-ext macro instead of the simple Dial(SIP SIP
SIP)
thank you
regards,
ron
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Re: [asterisk-users] simultaneous dial macro
hi,
thanks for your reply. is dialgroup already available in asterisk 1.4?
i'm currently using 1.4.21.
regards,
ron
--- On Mon, 7/28/08, Pavel Jezek [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Subject: Re: [asterisk-users] simultaneous dial macro
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Monday, July 28, 2008, 7:52 PM
you can try to place your macro extensions into single dialgroup using
DIALGROUP() function and then reference that dialgroup in dial aplication,
eg.
Set(DIALGROUP(test,add)=Local/100)
Set(DIALGROUP(test,add)=Local/101)
Dial(${DIALGROUP(test)})
ronald ramos wrote:
Hi,
Would just like to know if it's possible to be able to call a macro at
the same time.
i use a macro to dial local extension to another extension.
exten = 100,Macro(dial-ext|SIP/100)
exten = 101,Macro(dial-ext|SIP/101)
but now i would like to use it on a simple ringgroup where it will ring
all extensions
e.g. exten = s,Dial(SIP/100SIP/101)
how can i make use of my dial-ext macro instead of the simple Dial(SIP
SIP SIP)
thank you
regards,
ron
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Re: [asterisk-users] simultaneous dial macro
New in Asterisk 1.6
ronald ramos wrote:
hi,
thanks for your reply. is dialgroup already available in asterisk 1.4?
i'm currently using 1.4.21.
regards,
ron
--- On Mon, 7/28/08, Pavel Jezek [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Subject: Re: [asterisk-users] simultaneous dial macro
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Monday, July 28, 2008, 7:52 PM
you can try to place your macro extensions into single dialgroup using
DIALGROUP() function and then reference that dialgroup in dial aplication,
eg.
Set(DIALGROUP(test,add)=Local/100)
Set(DIALGROUP(test,add)=Local/101)
Dial(${DIALGROUP(test)})
ronald ramos wrote:
Hi,
Would just like to know if it's possible to be able to call a macro at
the same time.
i use a macro to dial local extension to another extension.
exten = 100,Macro(dial-ext|SIP/100)
exten = 101,Macro(dial-ext|SIP/101)
but now i would like to use it on a simple ringgroup where it will ring
all extensions
e.g. exten = s,Dial(SIP/100SIP/101)
how can i make use of my dial-ext macro instead of the simple Dial(SIP
SIP SIP)
thank you
regards,
ron
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[asterisk-users] IVR Direct Dial Extension
Hi, How can i enable the if you know your parties extensions please dial it now function? what do i need to add below? [ivr-1] exten = s,1,Answer exten = s,n,Wait(1) exten = s,n(begin),Set(TIMEOUT(digit)=3) exten = s,n,Set(TIMEOUT(response)=10) exten = s,n,Background(custom/myivr) exten = s,n,WaitExten(,) exten = hang,1,Playback(vm-goodbye) exten = hang,n,Hangup exten = 0,n,Goto(ivr-2,s,1) exten = 1,n,Goto(ivr-3,s,1) exten = 2,n,Goto(ivr-3,s,1) exten = t,n,Goto(operator-ext,100,1) regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote Support
Does anyone have any suggestions on what to use to monitor a vendor doing remote support? On the windows side things are typically done via screen sharing ( gotoassist.com, bomgar or similar) so at least you can see what the other end is doing. In working with linux (especially hardware vendors for asterisk) they want ssh root access, but I'm nervous about giving someone free range to a box without any type of monitoring. What does everyone else do? (besides not give them access). Looking at something like screen sharing or recording, perhaps keystroke logging. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Support
Take a look at 'screen'. Chances are, it's already installed on your boxen. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Joe Pukepail [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 28, 2008 5:02:58 PM GMT -06:00 US/Canada Central Subject: [asterisk-users] Remote Support Does anyone have any suggestions on what to use to monitor a vendor doing remote support? On the windows side things are typically done via screen sharing ( gotoassist.com , bomgar or similar) so at least you can see what the other end is doing. In working with linux (especially hardware vendors for asterisk) they want ssh root access, but I'm nervous about giving someone free range to a box without any type of monitoring. What does everyone else do? (besides not give them access). Looking at something like screen sharing or recording, perhaps keystroke logging. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] imap voicemail is being sent to the wrong imap account
I am testing the imap voicemail funtionality. I compiles asterisk using version 1.4.21.2 on rhel5.1. I have two different customers provisioned on the same asterisk as follows: taken from voicemail.conf: imapserver=192.168.196.43 imapflags=notls authuser=asterisk authpassword=asterisk ; Voicemail for customer '[EMAIL PROTECTED]' [cust1] 100 = 1234,Mike Oliveras,,,tz=pacific|[EMAIL PROTECTED] 200 = 1234,Grand Stream,,,tz=pacific|[EMAIL PROTECTED] 300 = ,Joe Blow,,,tz=pacific|[EMAIL PROTECTED] 400 = ,sipp,,,tz=pacific|[EMAIL PROTECTED] 800 = 1234,Michael Oliveras,,,tz=pacific|[EMAIL PROTECTED] ; Voicemail for customer '[EMAIL PROTECTED]' [cust2] 100 = 1234,Link Sys,,,tz=pacific|[EMAIL PROTECTED] 600 = ,Joe Blow,,,tz=pacific|[EMAIL PROTECTED] 700 = ,Fax Line,,,tz=pacific|[EMAIL PROTECTED] I am using dovecot as the imap server. If extension 200 calls extension 100 (both belong to cust1), then the voicemail is left in the mailbox belonging to cust2. Also, vmwi is sent to both [EMAIL PROTECTED] and [EMAIL PROTECTED] If user [EMAIL PROTECTED] checks voicemail, the response is that there are no messages waiting. If [EMAIL PROTECTED] checks voicemail, it is successful and vmwi is removed from both phones. I can send additional debug if needed, but I just wanted to check first if tere is any known reason why this would not work. I also tried removing the authuser and authpassword from voicemail.conf and provisioned an imappassword for each user and got exactly the same result. Is there any reason why I can't use the same mailbox number? This works fine when voicemail is stored locally. Best Regards, Mike Oliveras ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to ask good questions (was: Re: announcement server using asterisk)
Al Baker schrieb: Quote Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Grüße, Philipp Kempgen Yes - There is also a lot of bogus, incorrect, crap. His question was fair, on-topic, politely asked and as such hardly deserves to be made fun off Making fun of it was not my intention. But: http://www.gerv.net/hacking/how-to-ask-good-questions/ : Before you even ask a question, first try to find the answer by: 4. Searching the Web http://www.catb.org/~esr/faqs/smart-questions.html#before : When you ask your question, display the fact that you have done these things first; this will help establish that you're not being a lazy sponge and wasting people's time. Better yet, display what you have learned from doing these things. We like answering questions for people who have demonstrated they can learn from the answers. So for example Are there any PRI cards for laptops? (PCMCIA slot) All I found were BRI cards. is a lot better than - Are there any ISDN cards for laptops? - Yes. PCMCIA slot? - Yes. - BRI? PRI? - PRI. - ... And last but not least I just _have_ to post things like Google, learn.to/quote etc. from time to time. Of course it doesn't help but it makes me feel better. :-) Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR Direct Dial Extension
Nhadie wrote:
Hi,
How can i enable the if you know your parties extensions please dial it
now function? what do i need to add below?
[ivr-1]
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n(begin),Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(custom/myivr)
exten = s,n,WaitExten(,)
exten = hang,1,Playback(vm-goodbye)
exten = hang,n,Hangup
exten = 0,n,Goto(ivr-2,s,1)
exten = 1,n,Goto(ivr-3,s,1)
exten = 2,n,Goto(ivr-3,s,1)
exten = t,n,Goto(operator-ext,100,1)
regards,
nhadie
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You need to add a pattern that matches your extension's setup (3 digit
extensions get _XXX, etc...) and then call a macro or subroutine that
will perform the standard extension dialing:
EXAMPLE:
exten = _XXX,1,Macro(std-exten,${EXTEN})
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
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Re: [asterisk-users] Remote Support
Joe Pukepail schrieb: Does anyone have any suggestions on what to use to monitor a vendor doing remote support? On the windows side things are typically done via screen sharing ( gotoassist.com, bomgar or similar) so at least you can see what the other end is doing. In working with linux (especially hardware vendors for asterisk) they want ssh root access, but I'm nervous about giving someone free range to a box without any type of monitoring. What does everyone else do? (besides not give them access). Looking at something like screen sharing or recording, perhaps keystroke logging. I would suggest screen ( http://en.wikipedia.org/wiki/GNU_Screen ). screen doesn't solve the security aspect of your question though. Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Support
Philipp Kempgen wrote: I would suggest screen ( http://en.wikipedia.org/wiki/GNU_Screen ). screen doesn't solve the security aspect of your question though. Grüße, Philipp Kempgen Actually, it could. What I've done before, is give out an unprivileged account on the box (or some intermediate gateway box). Once they log in, you ask them to run screen (as the unprivileged user) to connect to a session you've created, then proceed to login as root yourself. If they disconnect their screen session, they leave your root terminal. You can also kill the screen session at any time. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow Playback of Recorded Files
Deric Page wrote: OS: CentOS 5.2 Asterisk: 1.4 I use NeoSpeech to do TTS recording. When I play back the files in Asterisk, the playback seems to be at about half-speed or less. However, when I play them through totem, xmms or other audio applications they sound fine. I’ve tried recording in 8-bit Mu-law PCM and OGG. Other formats I have tried (16-bit linear PCM, 8-bit alaw, 16-bit linear PCM wave, 8-bit unsigned linear wave, etc) haven’t played well at all (either not playing or producing static). Is there a generally known cause for slow audio playback? Thanks, -- Deric Page [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users sounds like the sample rate is wrong somehow -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Support
On Jul 28, 2008, at 5:50 PM, Jason Parker wrote: Philipp Kempgen wrote: I would suggest screen ( http://en.wikipedia.org/wiki/GNU_Screen ). screen doesn't solve the security aspect of your question though. Grüße, Philipp Kempgen Actually, it could. What I've done before, is give out an unprivileged account on the box (or some intermediate gateway box). Once they log in, you ask them to run screen (as the unprivileged user) to connect to a session you've created, then proceed to login as root yourself. If they disconnect their screen session, they leave your root terminal. You can also kill the screen session at any time. _ If you have X running you could also do VNC which would let you see what they are doing. Perhaps just change run level when they need access? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie in China: Red alaram in Zaptel for E1
This time, I am trying to remotely install Asterisk in China. I was told that an E1 line has been installed and so I plug it into port 1 of a TE412P. On the box, first of all, I just installed Zaptel 1.4.10.1. # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] # vi zaptel.conf [...] span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 *** However, I received a red alarm in zttool and the LED on the TE412P card is also red. *** I have made sure that the jumper is closed for port 1 on the TE412P card and so it could not be the jumper problem. ### Because this is the first time I install Asterisk in China and I was wondering if their E1 is different from the Euro E1. ### However, I went into dmesg and I discovered the following. ### Could it really be a zaptel bug? I saw on a similar few on the digium bug list but I cannot be 100% sure. Any thoughts? About to enter spanconfig! Done with spanconfig! Registered tone zone 33 (China) About to enter startup! TE4XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 128 channels BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681] Pid: 4681, comm:ztcfg EIP: 0060:[f8cba1df] CPU: 2 EIP is at init_vpm450m+0x32d/0x34a [wct4xxp] EFLAGS: 0286Tainted: G (2.6.18-92.1.6.el5 #1) EAX: EBX: f76ae8f0 ECX: 0019 EDX: ESI: f7997400 EDI: 0286 EBP: f76838c0 DS: 007b ES: 007b CR0: 8005003b CR2: b7f01007 CR3: 37bc1000 CR4: 06d0 [f8ca1b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8ca5ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042609c] release_console_sem+0x17e/0x1b8 [c046d53a] cache_alloc_refill+0x14b/0x450 [f8956f61] zt_ioctl+0x273/0x144f [zaptel] [c04d7d45] generic_make_request+0x248/0x258 [c045ae3c] __do_page_cache_readahead+0x69/0x1c6 [c0484a5b] __d_lookup+0x98/0xdb [c047c110] do_lookup+0x53/0x166 [c047e7e4] do_path_lookup+0x20e/0x25e [c047c389] permission+0xa2/0xb5 [c04e2d06] kobject_get+0xf/0x13 [c046f7fa] __dentry_open+0xea/0x1ab [c046f91f] nameidata_to_filp+0x19/0x28 [c046f959] do_filp_open+0x2b/0x31 [c048029b] do_ioctl+0x47/0x5d [c04804fb] vfs_ioctl+0x24a/0x25c [c0471bbe] __fput+0x13f/0x167 [c0480555] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Completed startup! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
My best guess from looking at that is that its a driver bug. The last thing that happens before the lockup seems to be an ioctl call to the device. Hope it helps, Igor H. Lee, John (Sydney) wrote: This time, I am trying to remotely install Asterisk in China. I was told that an E1 line has been installed and so I plug it into port 1 of a TE412P. On the box, first of all, I just installed Zaptel 1.4.10.1. # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] # vi zaptel.conf [...] span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 *** However, I received a red alarm in zttool and the LED on the TE412P card is also red. *** I have made sure that the jumper is closed for port 1 on the TE412P card and so it could not be the jumper problem. ### Because this is the first time I install Asterisk in China and I was wondering if their E1 is different from the Euro E1. ### However, I went into dmesg and I discovered the following. ### Could it really be a zaptel bug? I saw on a similar few on the digium bug list but I cannot be 100% sure. Any thoughts? About to enter spanconfig! Done with spanconfig! Registered tone zone 33 (China) About to enter startup! TE4XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 128 channels BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681] Pid: 4681, comm:ztcfg EIP: 0060:[f8cba1df] CPU: 2 EIP is at init_vpm450m+0x32d/0x34a [wct4xxp] EFLAGS: 0286Tainted: G (2.6.18-92.1.6.el5 #1) EAX: EBX: f76ae8f0 ECX: 0019 EDX: ESI: f7997400 EDI: 0286 EBP: f76838c0 DS: 007b ES: 007b CR0: 8005003b CR2: b7f01007 CR3: 37bc1000 CR4: 06d0 [f8ca1b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8ca5ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042609c] release_console_sem+0x17e/0x1b8 [c046d53a] cache_alloc_refill+0x14b/0x450 [f8956f61] zt_ioctl+0x273/0x144f [zaptel] [c04d7d45] generic_make_request+0x248/0x258 [c045ae3c] __do_page_cache_readahead+0x69/0x1c6 [c0484a5b] __d_lookup+0x98/0xdb [c047c110] do_lookup+0x53/0x166 [c047e7e4] do_path_lookup+0x20e/0x25e [c047c389] permission+0xa2/0xb5 [c04e2d06] kobject_get+0xf/0x13 [c046f7fa] __dentry_open+0xea/0x1ab [c046f91f] nameidata_to_filp+0x19/0x28 [c046f959] do_filp_open+0x2b/0x31 [c048029b] do_ioctl+0x47/0x5d [c04804fb] vfs_ioctl+0x24a/0x25c [c0471bbe] __fput+0x13f/0x167 [c0480555] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Completed startup! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
My best guess from looking at that is that its a driver bug. The last thing that happens before the lockup seems to be an ioctl call to the device. Hope it helps, Igor H. Thanks Igor. Does it mean that I should install a later release of zaptel? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
On Monday 28 July 2008 22:48:26 Lee, John (Sydney) wrote: This time, I am trying to remotely install Asterisk in China. I was told that an E1 line has been installed and so I plug it into port 1 of a TE412P. Are you sure that they're plugged into port 1 and not port 4? It is a rather common mistake to believe that the port numbers start at the bottom of the card and not at the top. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way voice after call transfer (bugs 9305, 13120)
Hello, I am having an issue here that after an attended call transfer there is no audio on one way; the problem is caused by Asterisk sending two INVITE messages without waiting for an ack for the first one. The issue has been reported on bug 9305, has been fixed and the fix is now included inside the main stream (version 1.4.21). However, I still get this behaviour, so I opened a new bug (13120). This bug sits there for over a week with no reponse... Has anyone else noticed this behaviour? Any idea what I can do? My users are angry on me... Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Are you sure that they're plugged into port 1 and not port 4? It is a rather common mistake to believe that the port numbers start at the bottom of the card and not at the top. Thanks Tilghman. I checked with the guys in the remote office and he is certain that he has plugged the E1 line into port 1. I am in the process of upgrading to zaptel 1.4.11. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
I think it can't hurt to try a different release. Let me know how it goes. Regards, Igor H. Lee, John (Sydney) wrote: My best guess from looking at that is that its a driver bug. The last thing that happens before the lockup seems to be an ioctl call to the device. Hope it helps, Igor H. Thanks Igor. Does it mean that I should install a later release of zaptel? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
