[Asterisk-Users] Re: [Asterisk-biz] Asterisk on Dell blade servers
One thing to be aware of is that Dell blade (as well as many other brand) servers are very heavy beasts. In any deployment with these, check the physical dimensions, check the weight and ensure that it will actually install into the rack that you are using. Also, check the power consumption and heat output and check with your data centre supplier once you know your final rack configuration that it is within their permitted limits. This is essential! Linus Magrathea - Original Message - From: Alistair Cunningham [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Commercial and Business-Oriented Asterisk Discussion asterisk-biz@lists.digium.com Sent: Tuesday, January 03, 2006 5:21 PM Subject: [Asterisk-biz] Asterisk on Dell blade servers We've been asked to quote for a large cluster running Asterisk and our ITSP in a box product. The system will be SIP throughout, with mixed codecs. We're considering using Dell blade servers, 1855 or similar, on the grounds that we normally use Dell machines and they work well, but we need higher rack density. Has anyone used these? Any feedback on whether they're good/bad/indifferent? What scalability do you get on simple SIP-SIP forwarding either with or without RTP passing through Asterisk? -- Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-biz] UK DID 0208 £1 per month
UK, London Based DID £1 per month All number begin with 0208 0xx Sam, Please, if you are going to market London numbers, format them correctly! The code for London is 020, therefore your numbers are 020 80xx . [Blatent self-plug] If you or anyone wants to purchase numbers from the rest of the UK we can offer DID/DDI from all UK area codes, *but* in wholesale qualities only. Linus Magrathea ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTCP-support
Please please please can any,all of you involved in this particular bug/patch please do whatever is required to get this into 1.2, whilst not directly affecting any of our internal configurations, does cause a number of support calls from Asterisk using clients who complain of dropped voicemail calls etc. http://bugs.digium.com/view.php?id=2863 Surely it can't be that hard for Asterisk to get one of the basic RTP features working! Linus - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: Users Asterisk asterisk-users@lists.digium.com Sent: Monday, October 03, 2005 8:46 AM Subject: [Asterisk-Users] *** Community alert :: Do you have open bugs inthe bug tracker? Asterisk buddies! If you have open issues in the bug tracker, please help us with providing fast responses. All developers are working real hard to close bugs pending the new release, so we kindly ask you for fast responses on our questions in the bug tracker. The quicker the better and we'll get 1.2 out of the door sooner. If you have new ideas, feature requests, thoughts - please keep the off the bug tracker until after the release. Thank you. Regards, /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Experiences with Termination Providers?
Indeed they do - but if you want numbers, you need to say where you are - there is no point our company supplying you with UK numbers or toll free, if you actually US people to call them! - Original Message - From: Me [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, November 30, 2004 11:48 PM Subject: Re: [Asterisk-Users] Experiences with Termination Providers? Mostly interested in US to US for now but interested in all areas, I was not aware I was restricted to looking for a provider in only certain areas. Most of the termination providers I have dealt with so far offer calling worldwide. Thanks, Todd -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: Linus Surguy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 28, 2004 9:34 AM Subject: Re: [Asterisk-Users] Experiences with Termination Providers? I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. It is an appropriate question - but I think the 'Welcome to the mailing list' message should point out that this is not a USA only list - anyone who posts this type of message should really say where they want service to and from! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + AS5300
Yes - all the recent IOS versions support SIP. - Original Message - From: Francisco [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 01, 2004 1:43 PM Subject: [Asterisk-Users] Asterisk + AS5300 Is it possible to terminate calls via SIP on a Cisco AS5300? Did anyone do it? How? Do i need an special IOS version? Ive been trying to compile the OpenH323 channel for the last month, but errors still happens. Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP-IAX-SIP silences
Hi all, We've got a number of users connected in a configuration which is basically: (a) SIP Phones - Asterisk - IAX - Our Asterisk - Cisco AS 5xxx (SIP) - PSTN We also have users in a configuration: (b) SIP Phones - Asterisk - IAX - Our Asterisk - Digium E1 - PSTN The second server in both cases is the same. The SIP phones place a call via these configurations to the PSTN Users in configuration (b) are happy. Some users in configuration (a) complain that x seconds into the call, the called party suddenly hears silence for y seconds, and then call corrects itself. Finally, we also have users in configuration (c): SIP Phones - Proxy - SER - Cisco AS 5 - PSTN These users are also perfectly happy. I suspect, given the history of SIP/IAX/SIP timestamps, something is being 'changed' which upsets the Cisco gateway resulting in the silences being sent. Any ideas or suggestions? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Experiences with Termination Providers?
I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. It is an appropriate question - but I think the 'Welcome to the mailing list' message should point out that this is not a USA only list - anyone who posts this type of message should really say where they want service to and from! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - how to get BT to present a number
We have a isdn30 line, with a DDI range. We also have 2 business units that have separate 0870 numbers that are mapped onto 2 DDI numbers. I would like to be able to present these 0870 numbers from the business units so that the correct number is displayed on a callerid, or when 1471 is dialled. BT claim that I can only have a single presentation number on the entire isdn line. Pipex claim that I can have a presentation number allocated to each DDI number that dials out. On the same ISDN line ! They simply take over the line, it's still fed into the BT exchange. Short of moving to Pipex, what can I do to get BT to allow me to do this ? Who do I need to speak to ? What techno speak must I give ? I'm afraid BT are 'correct'. Their presentation number product only allows one presentation number per subscriber line, therefore one per ISDN line and not one per DDI. However, alternative operators can do it, it is not a technical restriction. Also, if you are interested and have a suitable IP connection, and you wanted to VoIP your calls to us - we could do it for you that way! Linus Magrathea ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unknown number CID on SIP phone
I recently raised an issue which has just been fixed - this should clear your problem as well: http://bugs.digium.com/bug_view_page.php?bug_id=0002910 - Original Message - From: Brian McCrary [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 22, 2004 3:41 PM Subject: [Asterisk-Users] Unknown number CID on SIP phone Hello, I'm a new Asterisk user and I hope I haven't missed something, but I can't seem to find an answer to this issue. I have a Cisco SIP gateway terminating calls into a 7960 phone. The issue I would like to fix is if I have an incoming call without an ANI, such as directly from my TDM phone switch, Asterisk says the call is coming from the IP address of the Cisco gateway, withough the dots, so if my gateway is at 10.0.0.1, Asterisk reports a call from 10001 instead of reporting Unknown, or simply not reproting anything at all. It looks like there was some dicussions about a caller ID translation table. Is something like that what would be needed? Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: Incorrect parsing of 'unavailable' caller-ID fromCisco gateway
Before I raise this as a bug, it appears that * incorrect sets and reads the caller-id field from incoming sip packets when a Cisco gateway doesnt send one. Actually, dug into this further, and its an issue with reading Remote-Party-ID headers from the Cisco in get_rpid_num, so I've raised a bug. http://bugs.digium.com/bug_view_page.php?bug_id=0002910 Thanks for the response, Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incorrect parsing of 'unavailable' caller-ID from Cisco gateway
Before I raise this as a bug, it appears that * incorrect sets and reads the caller-id field from incoming sip packets when a Cisco gateway doesnt send one. For example, in this case the caller-id (PSTN) was unavailable, so the resulting SIP packet was: From: anonymous sip:1.2.3.4;tag=30A Asterisk reads, and sets and passes this out as a caller-ID of '1234' I would have thought Asterisk should have noticed that there wasnt a username portion present, and defaulted to its standard 'unknown' setting rather than trying to make something out of the hostname portion. Any comments? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Over 10,000 lines. Will asterisk manage?
4. Fiber will run to the main Telecommunication provider(PSTN) and 2 mobile providers. [snip] Keep in mind that their is no need for T1/PRI or any other type of external lines. Asterisk is to switch the voice data only. How are you linking to the PSTN referenced in (4) above then? How many concurrent calls have to go to the PSTN? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting CallerID on UK BRI line
yep, my mobile displays caller id for other numbers - and it even works perfectly displaying caller id information set by a cheap ISDN pbx on the *same* ISDN line as the Asterisk box. Curious. Even without setting a callerid on the outgoing calls I get No Caller ID on my mobile (or other phones - including other BT lines). BT are not withholding a number and I can change the callerID presented on the other phone system and it works perfectly. Strange, I will investigate more later. Remembering that you can only set caller ID to numbers that have been issued to you by BT, assuming you have a valid telephone number of, for example, 0118 321 1234 Try: SetCallerID(4) SetCallerID(34) SetCallerID(234) SetCallerID(11234) SetCallerID(211234) Normally you find it's either the 6 digit version or the single digit version that works with BT. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unusual protocols
examples of things which I have actually been asked about. There are a number of protocols based in 2600Hz tones (most US) and 2280Hz tones (mostly Europe), which are probably still spread quite widely in low density point-to-point connections. If there is anything you need, please tell me about it. I want to build a picture of what might be worthwhile tackling. You probably won't go far wrong by looking at the support offered by www.aculab.com and trying to match it . Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Database of world area codes
Also, try www.wtng.info Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] English vs American voice files
I've spent the afternoon recording all the files for the English speaking VM etc. I've parked the file here http://www.g7ltt.com/VoIP/vmukmale.tgz I did it with Audacity at 44.1KHz x 16bit and thenused sox to raise the levels to -3db and then again to down sample them into 8KHz GSM files. The few that I've listened to sound fine. Hi Mark, If you're going to publish these for public use it would be great if you could make them available in two versions, both the Asterisk 'standard' .gsm format, but also either in 8KHz/8bit/alaw raw or wav and/or 32Kbit ADPCM format - these do give a noticable increase in quality for local/PSTN users of telephony applications over GSM format. Either that, or if you could make the original 44.1K 16bit masters available so others could create the alternatives. Unfortunatly *'s ability to play these cleanly seems a bit broken at the moment, but at least we'll have them for when its fixed! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] English vs American voice files
1) One of the recordings says please enter the full 10 digit number starting with the area code. Any opinions on whether this should be changed for the UK and, if so, to what? Whilst you might be targeting the UK, it is still best to keep it generic - my suggestion would be simply 'please enter the full telephone number including the area code' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] English vs American voice files
rant Especially when asked to press pound! Pound! This is a pound £ not this # rant-end Mark, I would be happy to help and am actively seeking a suitable female, and my father speaks taff ! English gentleman seeks female for oral project? Hmmm...! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS Asterisk - an explanation
Actually, he meant that you need to have caller ID presentation turned on on the line so that when people call you, you can see their caller ID. However, I don't believe that that is a requirement for the sending side to work, just for the receiving side. I know for a while BT did say at fixed line SMS would not be available from ISDN30 lines, I don't know if this is still true. Linus - Original Message - From: Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, September 01, 2004 8:32 AM Subject: Re: [Asterisk-Users] SMS Asterisk - an explanation Thanks, but that was something I'd already checked. If I make a call out from that line to my mobile, then the number comes up as expected. Julian. - Original Message - From: Tim Robinson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, September 01, 2004 8:27 AM Subject: Re: [Asterisk-Users] SMS Asterisk - an explanation You need to make sure you have caller ID enabled on the line, as SMS relies on this to make it work. Rgds Tim Asterisk wrote: I tried to send sms messages the other day from a * box connected to a E1 line (BT ISDN30). Message never arrived, however, I was soon called back on the E1 by an automated BT system which sent a message stating that you cannot send sms messages on this line Is there anything I need to do before I start sending text messages ? Is it the ISDN30 that is the problem, and do I need to send SMS via standard lines (pots) or ISDN2e lines ? Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote: If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. I notice that the timing measurements are still showing wild values at times - here is a partial grab of an iax2 show channels: Lag Jitter JitBuf Format 00020ms 6291456ms ms ALAW 00012ms 6291440ms ms ALAW 00017ms 0004ms ms ALAW 00012ms 286523393ms ms ALAW 00012ms 0025ms ms ALAW -978714621ms 6293280ms ms ALAW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
At 17:10 29/08/2004, you wrote: I notice that the timing measurements are still showing wild values at times - here is a partial grab of an iax2 show channels: Lag Jitter JitBuf Format 00020ms 6291456ms ms ALAW 00012ms 6291440ms ms ALAW 00017ms 0004ms ms ALAW 00012ms 286523393ms ms ALAW 00012ms 0025ms ms ALAW -978714621ms 6293280ms ms ALAW Those wild times especially occur before any audio is sent. (e.g. while ringing or pre ringing). That maybe true, but the examples above appeared to be established calls! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice on BT ISDN Services (UK)
OK. You need one of the following: Home Highway Business Highway ISDN2e I can confirm that * works happily with all three - my office lines are (for various reasons, none of which apply any more!) on Business Highway. Heh, good old BT. I've never tested voice over Business Highway, as every BT engineer/support/sales person I've spoken to swore blind that it wouldn't work - and in BT's eyes, if they say it won't work, it's unsupported, therefore, if it breaks - you're on your own. Also, I don't believe you can get the full range of 'BT Select Services' or whatever they call them today on the Highway lines (things like Call Deflection, and even caller id on the home highway lines, I believe) BT fully support voice on Business Highway, they just assume that most people will use it for data. Most of BT's 'Digital Select Services' are not available on Home Highway, but are available on Business Highway. As I recall the only service not available on Business Highway, but is available on ISDN 2e is DDI across multiple 2B lines. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice on BT ISDN Services (UK)
Well I've just called BT, the confirmed to me that MSNs can only work with PTMP and DDIs with PTP. As for the seqential MSN issue, they have Complete tosh! As I said earlier, we've got it - and have ordered it with additional lines as well, if you really want it, just argue more, and talk to a specialist if required. However, Is there any reason (other than cost) to use MSNs over DDIs or the other way round? There is no real reason, but with ISDN2e you will be able to spread the DDI across multiple lines, something they won't do with MSN. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice on BT ISDN Services (UK)
OK. You need one of the following: Home Highway Business Highway ISDN2e I can confirm that * works happily with all three - my office lines are (for various reasons, none of which apply any more!) on Business Highway. If you want to use a Fritz! card, then you MUST have the line set up in PTMP (point to multi-point) mode. Home/Business Highway only work this way, so that's not a problem, but ISDN2e can be either PTP (point to point) or PTMP. BT PTMP only supports MSNs (which you can have a maximum of 8). PTP only supports DDIs (which you can have as many as you can afford). Essentially, This isnt actually at all correct, we certainly have a Business Highway line in PTP mode with MSN! (Although you are right in that this is the default) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P and Colt Telecom (Europe)
From the quote bits below: zaptel.conf span=1,0,0,ccs,hdb3,crc4 Assuming that it is the only E1 present, or the only one connected with the outside world, you should have the timing source configured: span=1,1,0,ccs,hdb3,crc4 Also, it might be that Colt are not using crc4 on your link, so try also with that removed: span=1,1,0,ccs,hdb3 Linus - Original Message - From: Aaron Clauson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 19, 2004 8:38 AM Subject: Re: [Asterisk-Users] E100P and Colt Telecom (Europe) Hi, Thanks a lot for the configs Fabe. I tried your zaptel.conf but I still get yellow and red alarms in zttool and * is unable to create any Zap channels (as expected with yellow and red alarms). I realise I will now have to start talking to Colt (in Ireland) to try and get the line up and running but if anyone has encountered this or something similar with Colt, or another provider in Europe, any tips would be greatly appreciated. Thanks, Aaron Message: 7 Date: Sat, 17 Jul 2004 10:38:26 +0200 From: Fabian Stelzer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] E100P and Colt Telecom (Europe) Reply-To: [EMAIL PROTECTED] zaptel.conf span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=nl zone=de doesen't work correctly for me :( but nl does... zapata.conf switchtype=euroisdn pridialplan=unknown signalling=pri_cpe group = 1 channel = 1-15,17-31 context=incoming this is the base config that works with colt... the rest has to be configured to you needs... Regards Fabe __ Do you Yahoo!? Vote for the stars of Yahoo!'s next ad campaign! http://advision.webevents.yahoo.com/yahoo/votelifeengine/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unavailable/Withheld identification
Speaking without experience of the exact combination you mention, but I'd expect that BT will send these to you using the combinations as follows: CLI: present Screening: available - Released number CLI: absent Screening: withheld - Withheld number CLI: absent Screening: available - Unavailable Number CLI: absent Screening: not available/interworking - Unavailable Number If you have access to the screening flag, this should help you. Linus - Original Message - From: Nick Barnes [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 19, 2004 2:07 PM Subject: [Asterisk-Users] Unavailable/Withheld identification Hi, I'm in the process of switching over to Asterisk from Alchemy kit and have hit a stumbling block. We're in the UK and use ISDN. At the moment we don't accept calls from withheld numbers (we just play them a message), but do accept calls from unavailable numbers. There doesn't seem to be any way for me to differentiate between the two number types in Asterisk (chan_CAPI) - they both appear to be presented as lacking callerID with no other identifier. I've had a look back through the archives and there doesn't seem to be an answer to this one. Does anybody have an idea on what to do or where to look? Many thanks, Nick. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get an UK SIP account with UK number?
network but say you have a local std code and you wanted to use that with a VoIP provider in the UK - in most instances you cant. Some are available to offer STD codes in certain towns but not all. It is possible to port geographic numbers between providers - I was able to port from BT to NTL. http://www.btwholesale.com/content/binaries/service_and_support/pricing_information/carrier_price_list_browsable/b1_08.rtf Tells you how much geographic number portability costs with BT, though you may also need to refer to other schedules if you intend to transit those calls over BT's network to wherever you connect to them. The only people who do it effectively have interconnects with BT in most exchanges, thus keeping these conveyance charges down. They would mostly be the cable operators, of course. We are in the process of establishing geographic portability arrangements in the UK for exactly this purpose. However, the paperwork involved is very time consuming to initiate a porting arrangement between one operator and another, and you have to do it for each and every operator you wish to port to/from. This is why most people are unable to offer it. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange * hangup issue
Anyone seen this one before? Asterisk Box (A): ATA186 SIP - Asterisk Asterisk Box (B) Asterisk - ZAP E1 Boxes linked by IAX2. ATA user places call to PSTN via box a b. Call fails and user hangsup. Box A CDRs reports call duration = 2 minutes Box B CDRs reports call duration = 3 hours How can this happen? I assume that if a IAX2 HANGUP is missed the protocol fully retries until done? Is it possible that HANGUP events do not retry as much as normal events? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Canadian DID
Can anyone point me in the direction of a wholesaler of Canadian DID numbers? If they'd be interested in trading them for UK numbering that would be even better! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AS5300 and Asterisk
Eric J Merkel wrote: Then you won't be able to us it for terminating voice. Sorry :( Eric Oh well, it was worth a though. Thanks for the answers though. I knew I should have gone for the 5350. If you get 'voice' cards for the AS5300 it'll work, and if you had gone for the 5350 you would still have to have purchased 'voice' licenses, rather than 'modem' licenses (and appropriate IOS) otherwise it still wouldnt have worked! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Time to lock down v1.1?
1.1 (today's head) is more of a let's try if this works' release. Please spend time testing it. Remember, CVS HEAD, is not meant to be stable. Now and then, it might not even compile cleanly. It's a developer's release, at some point in future aimed to be stable. Surely this is the reason of most peoples complaints today, all of us who are using Asterisk in real world, commercial environments get extremely frustrated when 'key' issues get fixed in the 'head' release, for example, recent fixes for IAX and SIP voice quality, and are not back ported to the stable/release/whatever version. It leaves us in a very difficult position, as commercially we are placing our users at unnecessary risk by using the 'head' version to get a specific bug fix, but also giving them poor service if we stick with the 'broken' version. Please can those responsible have some understanding of this, as a rule would it not make sense that all (or at least all major) 'fixes' go into both after being appropriately tested, and keep 'head' for the more 'bleeding edge' new features and more radical changes etc? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK ISDN PRI Problems
Hi Chris, I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which is currently working happily with an SDX Index phone system. I have to I can't see particular problems with your config, but I have a few comments: [snip] I have a Digium E100p card which is configured in zaptel.conf thus; span=1,0,5,ccs,hdb3,crc4 You do need to ensure you have networking timing otherwise you'll get frameslips. I don't know if the 'LBO' really has much effect here, so I would go for span=1,1,0,ccs,hdb3,crc4 Now, do you now you have CRC4 on the link? If you get this wrong, this will cause things not to work, and even if they do, you'll get noise on the link. Try without. span=1,1,0,ccs,hdb3 The zapata.conf is like this; [channels] [snip] rxgain=-5% txgain=+5% Whilst I know other people have posted differently, I can't see any legitmate reason for specifying gain on an end to end digital link. I'd suggest these should be zero. further red alerts. NTL, bless them, came out with a test rig and plugged this in the back of my * box and we made a series of test calls which all worked fine, although the NTL chap said the attenuation was out as there was a lot of buzz on the line. He suggested we set the line Sounds like pants to me, but what sort of 'buzz' are you talking about? Remember this is a digital link and not subject to analogue style interference. Many thanks for any help offered :) My money is on the CRC4 setting. Try that first. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANI II/Payphone indication
So.. Maybe I can get it on PRI by having it appended to the DNIS (as per Ryan Tucker's reply).. or maybe through some other kind of T1 arrangement that isn't PRI? Excuse my ignorance here but back in Europe it's pretty much primary rate ISDN or nothing, the whole 17 flavours of T1 is a bit of an unknown.. You're lucky that time has caused some standardisation, it was only a few years ago that here in Europe we had to deal with PRI in forms of DASS/2, Euro-ISDN, DPNSS ,1TR6, VN5... etc etc... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan
My dialplan is for the outgoing SIP call is:
exten = _00.,1,AbsoluteTimeout(3600)
exten = _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
exten = _00.,3,Answer
exten = _00.,4,Hangup
exten = _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
exten = _00.,104,Answer
exten = _00.,105,Hangup
I can't help with presenting busy to the SIP devices, but if you have the
above on any sort of PSTN gateway you are going to annoy the PSTN users - as
if the number selected is busy or otherwise unavailable you will still
'Answer' the PSTN call, causing the person calling to pay whatever call
establishment charges/minimum charges appropriate to their tariff.
Linus
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Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan
exten = _00.,1,AbsoluteTimeout(3600)
exten = _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
exten = _00.,3,Answer
exten = _00.,4,Hangup
I can't help with presenting busy to the SIP devices, but if you have
the
above on any sort of PSTN gateway you are going to annoy the PSTN
users - as
if the number selected is busy or otherwise unavailable you will still
'Answer' the PSTN call, causing the person calling to pay whatever call
establishment charges/minimum charges appropriate to their tariff.
Thanks for pointing that out.
Luckily Asterisk has a 'billed seconds' field in the cdr which is 0 when a
number is unavailable or busy despite showing the call as 'answered'.
A view could be taken that 0 length billed seconds calls need not be
billed with a minimum connection charge... perhaps.
Not ideal though.
Thats not quite the point, I was saying that if instead of this being SIP -
Asterisk, this was PSTN - Asterisk then you would have cost the caller real
money, as their telco, BT, ATT or whoever would have charged the caller as
a result of your 'Answer'. Asterisk's own billing records are not relevant
to this.
Linus
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Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan
Linus, I assuming that for incoming service something like . exten = incomingnumber,1,AbsoluteTimeout(3600) exten = incomingnumber,2,Dial(SIP/sipdevice,120) (maybe with ,r) exten = incomingnumber,3,Congestion exten = incomingnumber,103,Busy [where incomingnumber is whatever the incoming service number is presented as.] would probably not cause someones telco to charge a customer if the call wasn't picked up after 120 seconds or was busy immediately as the only possible 'answer' would be generated by the dialled device picking up? Exactly! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Different UK Caller ID question!
Can a standard BT phone that supports CID (Such as a BT Decor 310) pick up the CID information that asterisk passes out to analog lines or would I have to get an analog phone with CID from the states? Most of the BT brand caller display phones support both the BT/UK standard means of transmitting caller ID before ring and the US method of transmitting it afterwards, so you should be OK with Asterisk. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Who has access numbers in the UK and Germany?
I need a few access numbers in the UK and Germany. Does anyone have this available right now? I need the incoming calls to be directed We do. I'll mail you off-list. Linus Magrathea ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Who has access numbers in the UK and Germany?
I'm afraid I'm just out on a family 'outing', can you give me an overview via email of what you are looking for ? - Original Message - From: Stephen Karrington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, April 09, 2004 12:07 PM Subject: [Asterisk-Users] Who has access numbers in the UK and Germany? Hello, I need a few access numbers in the UK and Germany. Does anyone have this available right now? I need the incoming calls to be directed through IP to one of my asterisk servers in Europe. Please contact me off the list if you want. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadBRI and UK ISDN2e
Just been back through BT order processing and told them to put Caller Display (as they call it) on the line, which they said they've done... getting fairly certain it's not a BT issue now :| It might not be a BT issue, but BT *dont* call it 'Caller Display' on ISDN lines, you want 'ISDN 2e Calling Line Identity Presentation' See: http://www.serviceview.bt.com/list/current/docs/Exch_Lines.boo/001319.htm for pricing info. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadBRI and UK ISDN2e
I've just been onto BT again, and it seems they did have CLIP on the line for a few minutes, but then they removed it again and put COLP on, which they then told me was the same as CLIP... monkeys :( Wait... I take that back.. calling BT a bunch of monkeys is insulting to monkeys. They then proceeded to tell me that I didn't need to purchase a presentation number service (that allows us to display our non-geographic numberto people with CLIP rather than our geographic one)... I could just tell my switch to send whatever callerID I wanted and it'd get displayed... BT ISDN2e - the phreakers delight :) You clearly have been speaking to a branch of BT that doesnt have the slightest clue. I can confirm that the above is definately NOT true. You can set the caller ID you send out, but only within the range of DDI/MSN numbers that has been assigned to your ISDN 2e line. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadBRI and UK ISDN2e
I've just got my nice shiny quadBRI card, and it seems to be working very well - except for one little issue - CallerID. The card is currently connected to an ISDN2e line in P2P mode, and an S0 adapter on our existing alcatel PBX. The S0 connection recieves callerID and displays it correctly - the 2e line doesn't, and BT have said that CLID was enabled on the line two days ago. Does anyone have any pointers on this? Just a quick check, is it connected to 'real' ISDN2e or a Business Highway ISDN port? If the later, make sure that BT have turned caller display on the ISDN port and not on one of the analogue ports - this is a common mistake they make when taking your order. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP G.723.1 G.729 Asterisk Passthrough
Ok, I give up! I've tried all the suggestions that I can find to make this work, but it still doesnt. We are trying to send calls using G723.1 from a Cisco AS5300, via Asterisk to Cisco ATA-186 (and similar) devices. We have no support in Asterisk loaded for G723.1, we are simply trying to get * to pass the media stream through without change - it doesnt have to generate/receive any audio itself. We can't use reinvites as the AS5300 is not accessable to the outside IP world. Anyone out there got this working? If so, what did you have to do to your config to make it work? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP G.723.1 G.729 Asterisk Passthrough
Ok, I give up! I've tried all the suggestions that I can find to make this work, but it still doesnt. We are trying to send calls using G723.1 from a Cisco AS5300, via Asterisk to Cisco ATA-186 (and similar) devices. To follow up my own post, I have now got this to work. It would appear that the key action that has to be taken is to place allow=G723.1 at the top of the list of codecs in the general section, i.e.: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=lowdelay disallow=all allow=G723.1 allow=alaw allow=ulaw allow=gsm Place it anywhere else, and it doesnt work - dropping the call immediately it answers. It doesnt appear to be required to actually mention the codec anywhere else in any other section. Now, the problem is that I don't want to answer any other [inward] call and present G723.1 as an option, but of course with it in the general section, this is exactly what I've specified! Any ideas how to get out of this one!? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone?
Well why not? Everyone has to eat at the end of the day! Is it worth considering setting up an asterisk-trading mailing list specifically for this purpose? But surely we'd all just end up trying to sell to each other that way! At least being on the main mailling list means that we have plenty of customers to prey on?! Hotlinks Internet Services offers Voip grade bandwidth on our Juniper powered ob: Magrathea offers A-Z IAX termination, origination blah blah blah blah. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] European Caller ID
I know for example that when people are calling us from the US or from London CID is not captured by France Télécom, because we have a free service here where you can call a number to see if you missed any calls. This will actually depend on the carrier used to place the call from the UK to France. Some carriers remove the caller ID on international calls, some still send it. You might find if you are called by someone different in the UK you do get to see it! Also, the 'exchange' based caller id readout systems tend not to read out international caller id, but there are cases where the caller display devices would do so. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DPNSS and Asterisk
Hi Just one question do any of the Digium T1/E1 cards do DPNSS signaling? No. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P UK PRI Configuration
We have 1 ISDN PRI (E1) provided by BT (UK) for incoming and outgoing calls. Incoming calls work fine and the are no alarms on back of card or in /proc/zaptel/1, but with outgoing calls, all numbers are rejected with the BT error The number you have dialed has not been recognized, please check and try again Assuming that you are actually dialling the numbers correctly with your extensions.conf / Dial commands, then you might need to experiment with: ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown:Unknown ; private:Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN pridialplan=national in /etc/asterisk/zapata.conf - this is most likely what is going wrong. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
are vendor/partners in the states and the UK is next. Also if you can reccomend any voice providers we can work with I will be in the UK next week. If you are looking for 'wholesale' voice in the UK you could give us a try, contact our sales team at [EMAIL PROTECTED] or feel free to email me off-list directly ([EMAIL PROTECTED]) Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
Does anyone know where I can find some more info on the VoIP laws in the EU? VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU parliament), last time they looked at it a few years ago it wasn't perceived to be entranched enough to worry about, I suspect this will change soon. In the UK Oftel put out a guide, which says if you're running VoIP services (i.e. back-end services, so maybe a SIP proxy/registration server or interconnection with the PSTN) you are a Communications Service Provider and covered by the same regulations as a traditional voice provider. Just to clarify this from a different direction, Oftel/Ofcom approach these things by say that they are 'technology neutral', i.e. as standard they don't care how the service is delivered, it is the service that is regulated and not the delivery mechanism. This means in theory the rules for VoIP are the same for copper, wireless, mobile etc. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
Just to clarify this from a different direction, Oftel/Ofcom approach these things by say that they are 'technology neutral', i.e. as standard they don't care how the service is delivered, it is the service that is regulated and not the delivery mechanism. This means in theory the rules for VoIP are the same for copper, wireless, mobile etc. As I understand it that is what the Ofcom VoB discussion next week is all about.. The standard line telco's have to be required to provide a service in an emegency eg during a power failure, but this is impossible for a VoIP provider sine the provider does not have control over the full path or the electricity supply.. That is only one example where VoIP cannot be regulated in the same way as standard telephone services.. Thats not completely true - UK regulations say that a standard POTS analogue phone line must work in the event of power failure, and the same is true for a single ISDN line installation, but nothing else is actually covered - if you have a PRI ISDN30 install it is actually your responsibily to make it work in a power failure condition by providing UPS etc - if you want to. Equally VoIP tends not to fall under this requirement. I think we can expect that the meeting next week is going to primarily concentrate on a) 999 emergency calling requirements and b) numbering issues. Whilst there may be some other coverage of PATS/non-PATS issues* I'm sure these will be the main focus. Linus * Other PATS issues are things like directory enquiries/operator assistance/providing directories/itemised billing/service for the blind etc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Address Separator hex b causes callerid rejection
I missed your earlier message, but to try and help: a) You are correct, the hex 'b' usage is only valid in the UK specific BT IUP SS#7 interconnect protocol and therefore is nothing to do with PRI usage whatsoever (or indeed ISUP SS#7). b) In q931.c these various flags can be set for outbound CLI (caller id), from user provided not verified, user provider verified, user provided verfication failed, and network provided. c) BT however will only accept CLI that you are authorised to send - whatever the state of the flags, this means that you can only send CLI that matches numbers that have been allocated to your PRI. If you are trying to do otherwise this will always fail. Linus - Original Message - From: bam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 02, 2004 12:02 PM Subject: Re: [Asterisk-Users] Address Separator hex b causes callerid rejection I've now twigged that this is an SS7 flag and is being set in our switch as a result of * passing the Network provided screening indicator to a value that is interpreted as untrusted. Is there a simple way of changing the default value for this? At 16:22 30/01/04, you wrote: I am having a little bit of a problem with BT rejecting my callerid values as they are prefixed by hex b. This indicates that the caller id is user provided and not verified. Does anyone know how I can control where this appears in the cli? The purpose of the separator is described below: 1 - PNO 006 section 2.4.19 c note states that the hex b denotes an address separator, to separate the part which is network provided from that which is user provided - This means that it separates the extension number from the rest of the number. 2 - PNO008 section 22.1.3.3 states that the hex b dependant on its position, denotes whether the screening indicator is user provided not verified, network provided or user provided verified and passed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Address Separator hex b causes callerid rejection
The problem is that the ISDN call to our switch has the screening indicator set to untrusted, to the switch sticks 0xb on the front of the CLI. BT then drop the CLI altogether. So I need to find a way of fiddling with the * presentation. Ahhh. so you have an interconnect switch then I take it! You need to set the q931.c value - PRES_ALLOWED_NETWORK_NUMBER which has a value of '3'. I think, although I've never tried this, you can actually call the application CallingPres with the value of 3 before making an outbound call. CallingPres(3) I think should do it - someone else might be able to advise better. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Address Separator hex b causes callerid rejection
The problem is that the ISDN call to our switch has the screening indicator set to untrusted, to the switch sticks 0xb on the front of the CLI. BT then drop the CLI altogether. Actually, just as a point, the 0xb doesnt mean 'untrusted' as such, it just means user provided. I'd be interested if you'd like to dial 0870 068 9001 and let me know the date and time of the call to see if we see the CLI you are trying to pass. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bluetooth discussions
IRC channel chatter says that there are some new developments with a cool presence trick that Mark has come up with for bluetooth devices. I know a bit about it, but I think the general population here would like to see some details if they're available. I don't know if this is what you are talking about, but I know of other experiments where a bluetooth enabled server allows bluetooth enabled mobile (cellular) phones to register and then carry two way calls over bluetooth rather than GSM. This would be a cool trick if Asterisk could do this too... Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T400P E400P second source
From: Scott Stingel [EMAIL PROTECTED] I understand that there also is a new board from Digium, the TE405P, which is like the 3.3v TE410P, but uses a 5-volt supply. Any ideas if this is actually shipping yet though? If not when? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: FWD Holiday Promotion: Free Calling to 8 Countries
I know this is OT for this list, but I havnt seen it mentioned here and in the spirit of 'open source' I thought this would be interesting for readers here: - Original Message - From: Jeff Pulver [EMAIL PROTECTED] Sent: Tuesday, December 23, 2003 11:28 PM Subject: [FWD] FWD Holiday Promotion: Free Calling to 8 Countries Hi There, In the spirit of the holiday season, from today until the end of the year, it is now possible to use Free World Dialup to place, for free, calls into: Australia, Canada, China, Germany, Israel, Italy, United States and the UK. Note: Mobile calls can only be placed to people in the USA and Canada. To place a call, dial: * [country code] number on Free World Dialup. For example: -- USA/Canada: *1 Australia: *61 China: *86 Germany: *49 Italy: *39 Israel: *972 UK: *44 -- My hope is that our promotion will help some families and friends stay in closer touch during this holiday season. Please feel free to let others know about this. I'd appreciate your help in spreading the word and sharing the holiday spirit. :-) Best regards, Jeff p.s. I'm still working on getting the FWD list formally restored. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip-info.org DNS seems broken
For the last few days I can not resolve voip-info.org from many DNS servers. It does resolve with some DNS servers but I suspect it may be related more to caching. I've alerted James of the problems. I haven't seen them myself, so its hard for me to track. The wiki has become a too valuable resource for this community to continue to have these kind of problems. I assure you that James have spent a lot of time trying to solve the DNS and performance problems. Any help tracking the problems down is appreciated! If it helps, I'm willing for our company to offer secondary DNS. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
On Thu, Dec 04, 2003 at 10:34:02PM -, Linus Surguy wrote: I don't want to criticize your idea, but you do have to consider certain points. Starting from (as has already been mentioned) the bandwidth of DS3 is far too much to reasonably shove down the PCI bus without data loss / excessive overheads. ??? a standard 32 bit 33MHz PCI bus has a maximum bandwidth of 133MBps == 1Gbps. a DS3 is 45Mbps. even if you pass the data over the bus 10 times, you're still only using up half the peak bandwidth. Forgive me, it was a late at night comment after drinking too much - I don't know what I was thinking! However, as a general principle, if the PC was to pull traffic off the card, switch it (i.e. reorder it) and put it back on, it would probably take some time to get over the various timeing and synchronisation issues that this would present. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
I am uncertain of PCI bus speed limits - too many conflicting reports are wedged into my head. However, the intent here is to dump calls out via VoIP and not simply switch between channels elsewhere on the DS3, so overcoming that limitation needs to be addressed (if it exists at all, as a follow-up post has countered) or some other non-PCI solution created. Ideally, I'd like to see TDM on DS3 in, IAX2 on ethernet out after some minimal call control through a context. I assume that you are suggesting G.711 over IAX2 so there is limited CPU resource used on codec? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
I don't want to criticize your idea, but you do have to consider certain points. Starting from (as has already been mentioned) the bandwidth of DS3 is far too much to reasonably shove down the PCI bus without data loss / excessive overheads. Thus a sensible approach would be one where the card performs the switching, (H100/H110 or otherwise), leaving the Asterisk unit to maybe handle signalling and call control only. You could go one further, and if you require 'voice' resource, to switch that onto the PCI bus as well for processing. The way I see this, the best implementation plan would actually be to take a standard DS3 card with a H110/H100 bus, and then look for a third party card which could switch timeslots on the H110/H100 bus to the PCI bus. This composite approach would allow a zero latency switching path, but still include the flexibility of Asterisk. However, considering the traffic volumes that you are talking about, is it really true to say that the traditional telco cards are astronomically priced, given the amount of revenue that can be generated per month on a DS3? Linus - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 04, 2003 8:06 PM Subject: [Asterisk-Users] Port density: DS3 cards? Obviously, there are no DS3 TDM cards that are currently compatible with Zap channels. (or are there?) Does anyone know of an inexpensive DS3 card that could perhaps be used with Asterisk if one were to try to port the Zap drivers to such a card? PCI, of course, would be the bus of choice. I think there are quite a few discouraging comments to be made on that question. Firstly, most companies that produce telecom hardware have silly overhead, and thus the price of their cards is astronomical. Secondly, most companies that produce telecom hardware are of the opinion that transcoding (compression) should be done via DSP's, which inflates the cost of the card significantly. Thirdly, most telecom hardware vendors would not consider allowing their drivers into the public domain if such development were to happen. I've talked to some parties (you know who you are) who have expressed some interest in building this type of interface, but a situation where I can actually put my hands on equipment is far better than speculative interest by those who have not even decided to go forward with design, no matter how interesting the end product sounds on the whiteboard. However, regardless of all these negatives, I'm interested in any vendors anyone can offer as a starting point. Please, don't pester me with comments like Why do you need 28 PRI's? or You'll never use that much capacity. Assume that I actually DO have that volume of traffic, and assume there are several dozen other people on this list (lurkers and active people) who have the same requirements, and assume there are hundreds more people out there who have the requirement but haven't considered Asterisk because DS3 isn't an option. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Ethereal plugin v0.3 is out
- Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 02, 2003 7:24 PM Subject: Re: [Asterisk-Users] IAX2 Ethereal plugin v0.3 is out Alastair Maw wrote: On 28/11/03 07:39, Olle E. Johansson wrote: The latest version of my Ethereal plugin for IAX2 is now available here: - http://almaw.com/ethereal-iax2-plugin-0.3.zip Could you please create a URL that is a bit more non-version-specific? http://almaw.com/etheral-iax2/ It now, inevitably, has a web site. :) http://www.voip-info.org/tiki-index.php?page=How+To+Debug+and+Troubleshoot+V OIP Updated. Except both links appear to be rubbish! I think you meant: http://almaw.com/ethereal-iax2/ Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP bandwidth management with linux CBQ
Hi all, Is anyone here using linux as a router and managing their VoIP traffic with CBQ ? If so, do you have any configs (tc scripts etc) to share? We've trying to ensure that all VoIP traffic is prioritised ahead of 'normal' traffic, and at the moment have setup two classes based on the TOS flags. It all seems to work perfectly, but it would be interesting to see what other people are doing. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) Maybe asterisk-install ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application CallingPres
On Wed, 2003-11-19 at 15:21, Olle E. Johansson wrote: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+CallingPres Could someone explain this applicatoin a bit more? I found the application in the Zap channel source, and a comment says something about PRI connections. What is the value specifying? This CVS commit should help. http://lists.digium.com/pipermail/asterisk-cvs/2003-October/000205.html Call presentation is about whether or not callerid is displayed, what callerid is displayed, and similar. The value itself is a means of giving information about the callers telephone number. libpri incorrectly processes this as one value, where in actual fact it is two different values, stored within the same byte. The values itself is defined as follows in ITU Q931 Presentation indicator (octet 3a) Bits 7 6 Meaning 0 0 Presentation allowed 0 1 Presentation restricted 1 0 Number not available due to interworking 1 1 Reserved Screening indicator (octet 3a) Bits 2 1 Meaning 0 0 User-provided, not screened 0 1 User-provided, verified and passed 1 0 User-provided, verified and failed 1 1 Network provided In essence, it says 'is the person who has been called allowed to see the callers number' and 'what authority was used to verify that this is a genuine number'. When I have a moment I'll look at contributing a tidyup of the handling of this value as it is something that is important to us as a telco using Asterisk in a carrier environment. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PBX (Asterisk) - Cellular Phone Network
Some of the older cell phones use to expose either a 2-wire or 4-wire interface via a connector on the phone (don't know about transmission I won't bother with any of that - purchase a Nokia Premicell (or other manufacturers similar item). This device takes a normal GSM SIM card and then presents a normal PSTN line interface - plug that into your normal Asterisk PSTN line card - job done. I don't have time to go digging for the link, but there is at least one vendor who offers a hardware solution with 24 DS0 GSM cards in a channel bank, which you'd then connect to a T100P card. In other words, it's 24 lines of cell phones in a 2u package, with minimal wiring hassle. Google should show you the way... Indeed there are - we looked at one of these for a different project, here is some details from a German vendor: http://www.teles-communication-systems.com/G1/G19500AN0_GSM_frame.html Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I hate to do this but..
I think that the card they have with 4 E1's / H323 and ethernet could be considered just as a MG seen from the '*' side and that might be the most easy way to go if you dont want go all the way and make a channel driver. My 1c on that issue. Freddi The Aculab VoIP card actually only supports 2E1. Also, it is possible to use an API to drive it to handle the RTP only, leaving SIP/h.323 whatever to the application. However, what you can't do with the Aculab side of things (and I have though about this myself) is use it as a replacement for the Digium cards as there is no way to get the voice onto the PCI bus (even with Prosody). I'm afraid I would consider using these cards instead of the Digium ones otherwise for two reasons: 1) Worldwide certification and approval 2) Worldwide protocol support. However, they don't work and thats all there is to it! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
I can't remember who it was, but someone on this list was aiming to compile a list. We certainly replied, offering UK rest of world IAX (and SIP) termination. If that project isnt happening, it would be a great idea if someone else wanted to take it up. Linus - Original Message - From: Steve Sobol [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 3:31 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) Low, Adam wrote: We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway. If you'd like to speak to an account representative please contact me personally by email. Hmmm, this information should be on a website somewhere... -- JustThe.net Internet New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
- Original Message - From: Asterisk online forums [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 5:48 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) Linus, We started this list on Forums : http://asterisk.xvoip.com So any body can post info about services, etc which are off-topic for this list.. Just a suggestion, but under 'Service Providers' why not start a subject 'Other Providers'? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: I hate to do this but..
I'm afraid I would consider using these cards instead of the Digium ones otherwise for two reasons: 1) Worldwide certification and approval 2) Worldwide protocol support. Extra...Extra...Read all about it. Digium wants the whole world to know that the quad span T1/E1/PRI card, the TE410P, has passed Telecom FCC, Euro, and Australian certifications. This milestone in our hardware development gives the TE410P a significant boost for deploying applications worldwide and allows it to compete with similar cards in the market. We passed the rigorous certifications last week and the paperwork must be completed to make it official, but the hard work is done and we are excited to announce this great news to our customers. EMC testing is next. Jeremy McNamara Jeremy, That was actually my posting you acredited to Freddi. I am *not* 'bashing' Digium etc., I think the cards are great, give good value for money and combined with Asterisk provide a good solution. However, we work and assist people deploying around the world and in many cases that is outside the those areas you mention, and therefore although the Digium coverage is great, in some cases we need more, both in terms of approval and protocol coverage. Aculab (and similar such as Dialogic etc.) have spent a lot of time, and a lot of money getting worldwide approval / protocol support. If for example we want to deploy a 'legal' box in for example Hong Kong - we can't use Digium's cards, so we use Aculab - this is why if someone made an Aculab driver for Asterisk this would assist a lot of people deploying 'real' 'telco approved' applications. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does anyone provide inbound UK numbers using IAX ?
We can. Feel free to contact me and let me know your requirement. Linus Magrathea - Original Message - From: nathan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 04, 2003 10:18 AM Subject: [Asterisk-Users] Does anyone provide inbound UK numbers using IAX ? Hi All, Is there anyone providing UK geographic numbers that can be terminated on Asterisk using IAX ? It must be a geographic number (eg. Start 01 or 02, not 08xx). I've tried the sipcall.co.uk service and it looks very good when using X-Lite but it will not work with Asterisk. Switching to IAX should also resolve issues around NAT - hurray! -Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
Finally, are my options for handsets limited to IP phones via Ethernet, or analogue phones via a channel bank (and then to another Digium E1/T1 card), or is there the possibilty to re-use proprietary handsets from a previous PBX? One option you might not have considered is connect your existing PBX to the back of Asterisk and thereby use it as a channel bank itself. Linus Magrathea Telecommunications (provider of IAX termination and origination services in the UK!) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
Q931 is the RJ45 version that you just plug in to the line card. Q931 describes the protocol and not the line presentation. However, you do want to ensure that you ask for Q931 as although DASS/2 is an ISDN protocol, it isnt the same as Euro-ISDN and not supported by Asterisk. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
It's only worth doing if you're going to route them directly to some other kit, though, so Asterisk support for ISDN2 hardware is largely irrelevant here. I don't quite understand what you mean by this - we want to terminate the ISDN30e ourselves, and have a couple of ISDN2s also there 'just in case'. I think the person who replied meant that if you are having the lines as backup in case of failure, you should also be considering failure of the Asterisk equipment and therefore the backup lines should route to a different solution than the ISDN30e / Asterisk one. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 16xE1 solution based on *
XVOIP network is lunched, get your +1 777 number today. [EMAIL PROTECTED] XVOIP is lunched? Is this what happens after one is breakfasted? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
On Sat, 4 Oct 2003, James Sharp wrote: Actually, if this was to be done, it might be an idea to do it with DNS, so client machines would just do Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the DNS system would resolve which machine is the correct target - no cleverness at all required at the client end, so implementation would be portable across all the other gnophones etc. Yup. That would be the way to do it. I'll contribute the DNS code for it. Isn't that what e.164 was invented for? It is that type of mechanism that enum uses and yes it was to solve a similar goal, but in this case you need a 'route server' type system - in particular as this is for IP routing of PSTN end points not on an IP network. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
Each server would update from a master SQL database at a predetermined time. This way all servers would be in sync. Ok I get it, so there is going to be some app or script that will update the DB from a central source, and an AGI written for Asterisk that will do the lookup in the DB that was synced to find the extension and the server that extension is connected to, and then initiate a connection directly with that server.. Actually, if this was to be done, it might be an idea to do it with DNS, so client machines would just do Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the DNS system would resolve which machine is the correct target - no cleverness at all required at the client end, so implementation would be portable across all the other gnophones etc. Sounds workable.. Now what about these free local PSTN calls?? not all countries have free local calls.. how do you think the billing will be organised? However, this is a huge problem and I can't see it working.! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream
A little off-topic, but does anyone know if it is possible to disable call waiting on the Grandstream phones? I can't find anything in the configuration that would allow this. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk friendly IAX/SIP wholesalers in Australia
its a fair question: does anyone know any? I'm afraid this doesnt answer your question and is a bit of a shameless plug, but we have just started offering IAX (and SIP) termination in the UK, so if this helps anyone out, please feel free to contact me. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialing codes..( You can help! )
I am trying to setup some LCR functions on my Asterisk box and have a cheap call provider that uses various different numbers for landlines and cell phone numbers in various countrys.. I am finding it difficult to find the various codes.. eg. UK Landline - +44[12]. UK Cell - +44[7]. You have to be careful to be sure exactly what you are trying to obtain, the UK for example is: 441,442 - Geographic numbering 443,444 - not in use 44500 - Freephone 445other - 'multimedia/corporate' numbering, not much in use. 446 - not in use 4470 - UK personal 'follow me' numbering 4476 - UK Paging 4477/78/79 - UK Mobile 44800,44808 - Freephone 448 - Other, general non-geographic numbering, often used for company main numbering, eg. 0870 449 - UK Premium rate However, this doesnt necessarily tie in with what your carrier(s) will be charging or routing on. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialing codes..( You can help! )
I'd be up for setting up some kind of website/database thing for collating all this information, just not sure of the value and if anyone else would be up for it/contributing data? Be cool to have though, and nice for customer bill presentation etc? Try www.numberingplans.com or www.numberplan.org - they are both commerical but have some information for free. Also you can look at http://www.wtng.info/ a free site, and the ITU site at http://www.itu.int/ITU-T/inr/nnp/index.html its all there if you know where to look! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ITFS VoIP
I'm looking for toll-free #'s in: Germany Australia United Kingdom [snip] that ring to a US based PSTN #. I've contacted people like QWest, XO, etc.. and their rates are extremely high ($1.74/min from the UK). We use MCI Worldcom Teleglobe for our ITFS needs (we're based in the UK) and they don't charge anything like this for us although we are getting 'carrier' prices - I don't know what their 'retail' is like. Is there a better way to do this that involves VoIP? Possibly we could do UK 0800 for you over IAX or SIP if you interested - contact me off list if you'd like. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Possible FAQ: IAX2 - SIP with G729 and no licence
Assuming I've got a setup where calls entering Asterisk on SIP leave on IAX2 ( and the reverse), i.e. a SIP user might dial '1234' where we then have extern = 1234,1,Dial(IAX2/somewhereelse) Now, we don't have any G.729 functionality on this server, so what happens if the SIP user calls with G.729 only available? Assuming the remote IAX2 server does have G.729 can it be passed through to it? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] echo cancelation
HI all, Having a mental block today - can someone confirm which direction the echo cancelation applies to for the Zap PRI channels? ie. is it removing traces of the transmitted data to the PSTN from the received data, *or* is it removing traces of the data transmitted to Asterisk from the data received back from Asterisk? Got a configuration that is based on a call made from the PSTN to a SIP user: PSTN - Telco Switch - PRI - Asterisk - SIP - X-Lite PSTN caller hears no echo X-Lite user hears significant (1 second or so) echo If we setup a call so that the 'telco switch' runs an IVR process instead of linking to the PSTN, keeping the call entirely in the digital domain so no signal reflection the X-Lite user does not hear echo. Echocancel=yes etc. is enabled in the zapata.conf Which party is at fault? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK Caller ID and X100p
Hi I really need caller id to work in the UK, I understand that the X100p uses a US chipset,two questions 1) is that a product that converts UK to US caller id in line Not really the answer you were looking for, but if you get a line from a non-BT supplier (e.g. NTL or Telewest) you are quite likely to find that CLI works. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to connect 2 TE410P
And if we are going to get carried away: RJ45 CAT5 CAT5e - Original Message - From: Thilo Salmon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 08, 2003 3:18 PM Subject: Re: [Asterisk-Users] how to connect 2 TE410P Jeremy, Great idea, let's add E1 and E-1 as well so people on this side of the world have a chance to find this entry. ;-) Thilo On Mon, 2003-09-08 at 11:37, Jeremy McNamara wrote: Thilo, you are right, but lets add some nice keywords, for the archives: T-1 cross over cable pin out wiring diagram Jeremy McNamara Thilo Salmon wrote: Kelvin, 1 - 4 2 - 5 4 - 1 5 - 2 does it for me. Thilo On Mon, 2003-09-08 at 07:22, Kelvin Chua wrote: neat! actually we are just in the process of planning for an asterisk based simulation lab for the university. do you have a cable pin-out descriptions for that purpose? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- [netzquadrat] GmbH | Thilo Salmon Ronsdorfer Str. 74 | Fon: +49 211 302033 12 40233 Duesseldorf| Fax: +49 211 302033 22 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Known problem?
Haha. Last time I discovered my libc or something wasnt quite there so I couldnt compile chan_enum or something, so I didnt bother and carried on with the older version (only from a couple of months back) Linus - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 11, 2003 2:53 PM Subject: Re: [Asterisk-Users] Known problem? what hassles?cvs update Jeremy McNamara Linus Surguy wrote: Hi all, We're using an older version of *, built a couple of months ago and before we go through all the hassle of updating source files and checking latest dependancies on other kernels etc, I'd like to know if the following is a known fault: We're running a PSTN to FWD gateway in the UK and just whilst I was looking at something else I noticed a call come in which caused Asterisk to simply halt, terminating all processes. I've got a SIP trace of the call, which is quoted below. Any ideas? voip-gw1:/etc/asterisk # asterisk voip-gw1:/etc/asterisk # asterisk -rvvv == Parsing '/etc/asterisk/asterisk.conf': Found Asterisk 0.4.0, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk 0.4.0 currently running on voip-gw1 (pid = 31349) -- Remote UNIX connection voip-gw1*CLI sip debug SIP Debugging Enabled voip-gw1*CLI iax2 no debug IAX2 Debugging Disabled -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 -- B-channel 3 successfully restarted on span 1 -- B-channel 4 successfully restarted on span 1 -- B-channel 5 successfully restarted on span 1 -- B-channel 6 successfully restarted on span 1 -- B-channel 7 successfully restarted on span 1 -- B-channel 8 successfully restarted on span 1 -- B-channel 9 successfully restarted on span 1 -- B-channel 10 successfully restarted on span 1 -- B-channel 11 successfully restarted on span 1 -- B-channel 12 successfully restarted on span 1 -- B-channel 13 successfully restarted on span 1 -- B-channel 14 successfully restarted on span 1 -- B-channel 15 successfully restarted on span 1 -- B-channel 17 successfully restarted on span 1 -- B-channel 18 successfully restarted on span 1 -- B-channel 19 successfully restarted on span 1 -- B-channel 20 successfully restarted on span 1 -- B-channel 21 successfully restarted on span 1 -- B-channel 22 successfully restarted on span 1 -- B-channel 23 successfully restarted on span 1 -- B-channel 24 successfully restarted on span 1 -- B-channel 25 successfully restarted on span 1 -- B-channel 26 successfully restarted on span 1 -- B-channel 27 successfully restarted on span 1 -- B-channel 28 successfully restarted on span 1 -- B-channel 29 successfully restarted on span 1 -- B-channel 30 successfully restarted on span 1 -- B-channel 31 successfully restarted on span 1 -- Executing Dial(Zap/3-1, Sip/[EMAIL PROTECTED]) in new stack -- Accepting call from '118900' to '099138269' on channel 3, span 1 Interface is eth0 IP Address is 213.166.5.129 We're at 213.166.5.129 port 2738 Answering with preferred capability 8 Answering with preferred capability 4 Answering with preferred capability 2 Answering with non-codec capability 1 10 headers, 11 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK3f7299f7 From: 118900 sip:[EMAIL PROTECTED];tag=as5770a04f To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 237 v=0 o=root 31365 31365 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 2738 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 192.246.69.223:5060 -- Called [EMAIL PROTECTED] Sip read: LI SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK3f7299f7 From: 118900 sip:[EMAIL PROTECTED];tag=as5770a04f To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: Free World Dialup (0.8.11pre31 (i386/linux)) Content-Length: 0 8 headers, 0 lines Sip read: LI SIP/2.0 302 MovedTemporarily Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK3f7299f7 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=as5770a04f To: edc-soft sip:[EMAIL PROTECTED];tag=16f2d190 CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5062;q=1.000 User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 Content-Length: 0 9 headers, 0 lines -- Got SIP response 302 MovedTemporarily back from 192.246.69.223 Transmitting
[Asterisk-Users] caller display in a telco environment.
Hi John,
In response to your question, and a couple of other items seen recently,
I've produced this, which is my view on the Q931/PRI caller display issues.
(Due to different terminology I'm using the UK abreviation CLI, caller line
identity in the following text, but it could also be termed caller display
or ANI or 'A' Number)
There have been a number of threads recently regarding CLI both entering *
from a PRI, or being delivered from * to a PRI and in most cases this is
handled 'incorrectly' if the equipment is to be used in a telco environment.
I'm not currently that famililar with the * code, but this should help those
that are make the correct choices when making patches.
Firstly, some definitions. Q931 specifies two seperate flags, which the
'zap' libraries incorrectly bundle into the one set of 'presentation'
#defines.
The first is the presentation indicator, which has three primary values:
0 which indicates that the CLI if present is freely available for display
purposes,
1 which indicates that the user/users system has made a decision not to make
the number available for display, or
2 which indicates that the number cannot be used for display, but this is
not because of user choice action. (although the title is 'number not
available due to interworking' this doesnt mean there is not a number)
(These are in bits 7 6 and so take the values in the field of
0x00,0x20,0x40,0x60)
The second is the screening indicator. These values are:
0 - The user equipment has provided the number and no network equipement has
attempted to verify the number. (In all these definitions, network equipment
refers to authoritive switching systems that are part of the PSTN and can be
trusted to provide genuine information).
1 - The user equipment has provided the number and network equipment has
validated it.
2 - The user equipment has provided the number and network equipment has
rejected it.
3 - Number has been provided by authoritive network equipment.
Now, dealing first with calls arriving from a PRI. As a telco, connections
are generally classified as trusted or not trusted. Connections to a user
are always not trusted, connections using a protocol that does not support
withholding number display are not trusted and international connections are
not trusted. As far as Asterisk is concerned sensible defaults would be to
count PRI as trusted and all none PRI (SIP, IAX etc) as none trusted, but
this should be a per trunk/user configuration option (trustwithcli=yes|no?).
Thus if Asterisk is delivering a call that arrives on a PRI it should only
pass CLI if the presentation indicator is zero unless the trunk is 'trusted'
in which case the CLI can be passed in all cases, with whatever flags the
outgoing protocol allows to closely mirror the PRI flags - especially the
presentation indicator.
Secondly, dealing with calls being delivered out from Asterisk on a PRI, to
be standards compliant, (unless the call also arrived on a PRI), all calls
should send the CLI provided but marked 'number not available due to
interworking', and 'user provided not screened' - a byte value of 0x40. A
second per user/trunk configuration item should be provided
(clivalid=yes|no?) which then overrides this behaviour and then allows the
number to be delivered to the PRI with the flags set as presentation
allowed, user provided not verified, a byte value of 0x00.
Whilst the above doesnt show any code, hopefully it is enough to help!
Linus
Magrathea
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Sent: Tuesday, August 05, 2003 7:25 PM
Subject: Re: [Asterisk-Users] Syntax for hiding caller ID but still passing
ANI?
Lorenzo -
I've submitted a feature request with this patch
(http://bugs.digium.com/bug_view_page.php?bug_id=052). Your
patch isn't completely descriptive, since I still don't know how you
set the hidecallerid value from within a dialplan. Can you explain a
bit more, please? Have you submitted a disclaimer to Digium so this
patch might be added if it's seen as a useful addition?
Linus -
Thanks for the specifications. Did you have a patch or comments on
how this might be implemented in the code?
JT
We did something like this in chan_zap at pri_call() time:
case SIG_PRI:
[...]
if (ast-callerid) {
strncpy(callerid, ast-callerid, sizeof(callerid)-1);
ast_callerid_parse(callerid, n, l);
if (l) {
ast_shrink_phone_number(l);
if (!ast_isphonenumber(l))
l = NULL;
}
}
[...]
if (l) {
pres = ast-hidecallerid ?
PRES_PROHIB_USER_NUMBER_NOT_SCREENED :
PRES_ALLOWED_USER
} else
pres = PRES_NUMBER_NOT_AVAILABLE;
if (pri_call(p-pri-pri, p-call,
p-digital ? PRI_TRANS_CAP_DIGITAL : PRI_TRANS_CAP_SPEEC
p-prioffset, p-pri-nodetype == PRI_NETWORK ? 0 : 1, 1, l,
p-pri-dialplan - 1,
c + p-stripmsd, p-pri-dialplan - 1,
Re: [Asterisk-Users] Syntax for hiding caller ID but still passing ANI?
It seems that there are two options that block the presentation: 0x23 and 0x21 No no! Protocol wise, they are different flags in each nibble. If the high nibble is not zero then the number is not for general use. Any number presentation block in PRI-SIP conversion should only pass the CLI if the high nibble is zero. Linus JT l is set a couple of lines above. Basically l carries the nubmer so if there is no callerid in 'l' then we send this other flag 'callerid not available'. You need to choose one of these flags: /* Presentation */ #define PRES_ALLOWED_USER_NUMBER_NOT_SCREENED 0x00 #define PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN 0x01 #define PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN 0x02 #define PRES_ALLOWED_NETWORK_NUMBER 0x03 #define PRES_PROHIB_USER_NUMBER_NOT_SCREENED0x20 #define PRES_PROHIB_USER_NUMBER_PASSED_SCREEN 0x21 #define PRES_PROHIB_USER_NUMBER_FAILED_SCREEN 0x22 #define PRES_PROHIB_NETWORK_NUMBER 0x23 #define PRES_NUMBER_NOT_AVAILABLE 0x43 I think it might be PROHIBPASSED_SCREEN but not sure. Check q931 specs. Martin On Mon, 4 Aug 2003, John Todd wrote: I have a question regarding the flags for hiding caller ID presentation: My customer has a requirement that they are able to specify if outbound calls (on a T100P) will have the caller ID displayed or not. This could be easily solved, of course, by not setting a caller ID when creating the outbound call. However, the PRI to which this T100P is connected _must_ see a valid caller ID, and the caller ID is used for billing purposes. I know that there is the ability to hide caller ID within the Zaptel libraries, using the presentation flags. If set correctly, the expected behavior would be that the ANI would be sent to the switch, but with a flag that would tell the remote switch to suppress the caller ID from being transmitted to the end user. How does one activate that presentation switch from within a dialplan? Searching the archives gives me some hints, but no answers. Searching the code, I see in channels/chan_zap.c around line 1399 that the PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN and PRES_NUMBER_NOT_AVAILABLE are referenced, but I'm not clear on where l is set, or even if that is a trigger. Can someone give me a hand on syntax on this? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Syntax for hiding caller ID but still passing ANI?
(as a follow up to my own post) It seems that there are two options that block the presentation: 0x23 and 0x21 No no! Protocol wise, they are different flags in each nibble. If the high nibble is not zero then the number is not for general use. Any number presentation block in PRI-SIP conversion should only pass the CLI if the high nibble is zero. Cut and paste from the ITU Spec Q931 Presentation indicator (octet 3a) Bits 7 6 Meaning 0 0 Presentation allowed 0 1 Presentation restricted 1 0 Number not available due to interworking 1 1 Reserved NOTE 1 - The meaning and the use of this field is defined in 3/Q.951 and 4/Q.951. Screening indicator (octet 3a) Bits 2 1 Meaning 0 0 User-provided, not screened 0 1 User-provided, verified and passed 1 0 User-provided, verified and failed 1 1 Network provided NOTE 2 - The meaning and the use of this field is defined in 3/Q.951 and 4/Q.951. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] the 'pound' and '#' are the same?
Ryan == Ryan Tucker [EMAIL PROTECTED] writes: Ryan They are the same key. I'm not sure how the # came to be associated Ryan with the word pound, but in American English at least, they're the Ryan same key. The weight measurement pound is abbreviated lb. # looks similar in some handwritings, thus the use of # for lb avoirdupois. It is an interesting coincidence then that the # on a US keyboard is in the same place as a pound(£) symbol on a UK keyboard! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'Echo' - I'm sure a common topic
We're currently running a PSTN - SIP gateway with Asterisk. We also run IAX/SIP - PSTN. We have performed a test where the call is routed UK PSTN - Digium E1 card - Asterisk GW - SIP G.711 - FWD - X-Ten softphone There is no echo at the softphone end, but severe echo on the PSTN side. We've also performed a test Its not perhaps as simple as acoustic echo on the softphone side heading back to the PSTN. IE - speakers and microphone? In which case, the user needs to get a headset... I did ask them to turn down the speakers and retest and it appears the echo was still there. I don't quite understand how the echo is being introduced in this case. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'Echo' - I'm sure a common topic
I'll try to set up a retest and report back. Linus Can you have them try with just a headset? Mark On Mon, 21 Jul 2003, Linus Surguy wrote: We're currently running a PSTN - SIP gateway with Asterisk. We also run IAX/SIP - PSTN. We have performed a test where the call is routed UK PSTN - Digium E1 card - Asterisk GW - SIP G.711 - FWD - X-Ten softphone There is no echo at the softphone end, but severe echo on the PSTN side. We've also performed a test Its not perhaps as simple as acoustic echo on the softphone side heading back to the PSTN. IE - speakers and microphone? In which case, the user needs to get a headset... I did ask them to turn down the speakers and retest and it appears the echo was still there. I don't quite understand how the echo is being introduced in this case. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK Gateway
We're in the process of testing some equipment and configurations and to do this we have setup a UK PSTN Gateway to Free World Dialup. Simply dial 0845 004 5566 (UK local rate call) and at the prompt enter the FWD subscriber number - within a couple of seconds you should be connected. We can also terminate UK 0800/0808 numbers for SIP/IAX - PSTN calls, at the moment we don't have an FWD number setup for this, but simply use the username/password of guest/guest and point your connection at voip-gw1.magrathea-telecom.co.uk and send the number as 0800xxx Of course, this is all a trial at the moment, so no commerical warranties are available! Finally, if you wanted to trial your own personal 0870 number pointing to FWD, drop me an e-mail and I'll tell you how to do it. Linus Magrathea ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
