Hello,
I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 17 January 2008 01:54 To: [email protected] Subject: asterisk-users Digest, Vol 42, Issue 58 *** WARNING *** This mail has originated outside your organization, either from an external partner or the Global Internet. Keep this in mind if you answer this message. Send asterisk-users mailing list submissions to [email protected] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. Re: [IAX] Up-to-date list of soft- and hardphones? (Gordon Henderson) 2. Re: Can DB() use SQLite instead of BerkeleyDB? (Tilghman Lesher) 3. Re: WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101' (Andrew Joakimsen) 4. Re: Digium Part#'s (Was: Difference between TE121 and TE122) (Kevin P. Fleming) 5. asterisk to mysql database! (Naveen Palani) 6. Re: asterisk to mysql database! (Simon Elliston Ball) 7. Asterisk 1.4.17 and RXFAX via T38 (Robert Moskowitz) 8. Re: Unable to open master device '/dev/zap/ctl' (Chris Bagnall) 9. Re: [IAX] Up-to-date list of soft- and hardphones? (Vincent) 10. Re: Can DB() use SQLite instead of BerkeleyDB? (Vincent) 11. Re: asterisk to mysql database! (Tilghman Lesher) 12. Re: [IAX] Up-to-date list of soft- and hardphones? (Tim H. Panton) 13. HDLC errors (Steven) 14. Re: HDLC errors (Russell Bryant) 15. AddQueueMember and Flash Operator Panel ([EMAIL PROTECTED]) 16. Re: HDLC errors (Steve Totaro) 17. Anyone Using a Dell PowerEdge T105 in Production (Steve Totaro) 18. Problem with a channel (Ruben Zamora) 19. Re: HDLC errors (Andrew Joakimsen) 20. IMAP client in asterisk not trying to contact IMAP server (KodaK) 21. Asterisk Now Beta 6 and CISCO IP 7910 ([EMAIL PROTECTED]) 22. Re: Anyone Using a Dell PowerEdge T105 in Production (Erik Anderson) 23. Re: Anyone Using a Dell PowerEdge T105 in Production (Steve Totaro) 24. Asterisk on ClarkConnect (shadowym) 25. Re: Unable to open master device '/dev/zap/ctl' (Walter Willis) 26. Re: Anyone Using a Dell PowerEdge T105 in Production (Erik Anderson) ---------------------------------------------------------------------- Message: 1 Date: Wed, 16 Jan 2008 18:08:23 +0000 (GMT) From: Gordon Henderson <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Wed, 16 Jan 2008, Vincent wrote: > Hello > > There's a lot of information on VoIP at www.voip-info.org ... > but there's also a lot of outdated information there as well :-/ > > Since SIP is a pain to use when NAT is involved, especially when both > the Asterisk server and the remote phones are behind NAT... I'd like > to try IAX to see how it works and if it solves the issue. > > I'd like to start with a softphone (Windows only), and then, if tests > prove successfully, buy a hardphone. What would be your > recommendations? IDEFISK or Zoiper as it's called now. However, you'll need to do similar things to your asterisk box & router if it's behind NAT for IAX as you do for SIP. (You will need a static IP address on the NAT router and port-forward 4569 to the asterisk box, just as you'd port-forward 5060 and 10000-20000 for SIP) And a SIP phone behind a NAT router is also solvable if it supports STUN. I know that SIP behind NAT isn't perfect, but with care, it's very usable and workable. I have many installations doing just this, as I'm sure many others on the list have too. Gordon ------------------------------ Message: 2 Date: Wed, 16 Jan 2008 12:10:35 -0600 From: Tilghman Lesher <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Can DB() use SQLite instead of BerkeleyDB? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" On Wednesday 16 January 2008 10:02:12 Vincent wrote: > Before I bother calling a PHP script through AGI just to read a number > and rewrite the CID name... I was wondering if Asterisk could be > configured so that DB() uses a SQL server instead of the usual > BerkeleyDB? No, it cannot. You could use func_odbc to formulate your own queries, though. -- Tilghman ------------------------------ Message: 3 Date: Wed, 16 Jan 2008 14:03:53 -0500 From: "Andrew Joakimsen" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101' To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=UTF-8 Did you look at the trace I send you in email? Because in each request there are two IN IP lines I think Asterisk should only interpret the first one, On Jan 16, 2008 2:40 AM, Johansson Olle E <[EMAIL PROTECTED]> wrote: > > 16 jan 2008 kl. 04.43 skrev Andrew Joakimsen: > > > Well can you offer some explanation why T.38 faxing worked for months > > and then one day stopped working? > You are asking the wrong forum. Your device is clearly sending a bad > SDP. Ask the vendor of that device. > > /O > > > > > > Using both Linksys & Audiocodes (yuck) ATA. The first second of the > > fax tone is heard and then the T.38 switchover is attempted and the > > call drops with said error. > > > > > > > > > On Jan 15, 2008 6:25 PM, Mark Michelson <[EMAIL PROTECTED]> wrote: > >> > >> Andrew Joakimsen wrote: > >>> Anyone else have issues with T.38 where the call drops after T.38 is > >>> attempted to be negotiated, with a message like the below? > >>> > >>> WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host > >>> in > >>> c= line, 'IN IP4 100101' > >> > >> The problem is that 100101 is neither a valid IPv4 address nor a > >> fully-qualified > >> domain name. > >> > >> _______________________________________________ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > --- > * Olle E Johansson - [EMAIL PROTECTED] > * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden > > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ------------------------------ Message: 4 Date: Wed, 16 Jan 2008 13:04:01 -0600 From: "Kevin P. Fleming" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Digium Part#'s (Was: Difference between TE121 and TE122) To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 Dave Fullerton wrote: > If you want to know what a card's capabilities are you're better off > just memorizing each part number. Maybe there's a scheme I'm just not > capable of understanding here. We gave up (intentionally) on trying to have model numbers that reflected all the capabilities of each card, because they would turn into unintelligible (and unmemorizable) part numbers. We now have 'part numbers' that represent a given card with the options it was ordered with (analog module(s), echo canceler, etc.), and we've stopped trying to use suffixes to indicate bus type and instead just use a different model number. This why the TE122 (which replaced the TE120P) no longer has a 'P' suffix; the PCI-Express version is a different model number entirely. With that said, for some reason our marketing department decided to change the *prefix* for PCI-Express analog cards from TDM to AEX, but they still follow the rest of the model naming scheme (no suffix letter and no different model numbers that indicate included optional modules). -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Asterisk Experience" (TM) ------------------------------ Message: 5 Date: Wed, 16 Jan 2008 13:11:26 -0600 From: Naveen Palani <[EMAIL PROTECTED]> Subject: [asterisk-users] asterisk to mysql database! To: <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="utf-8" Hello, Is there a possibility to connect from asterisk to mysql database without the interface application like Ruby or PHP. If i can connect to mysql database from asterisk, i can update the database for manipulations. Appreciate your response. Regards, Naveen.Palani ________________________________ ?Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, ISO?9001:2000 assessed delivery processes and provides solutions in areas of E-Business, ERP, Application Management Services, and EAI to customers in BFSI, Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global Delivery Model.? This e-mail and any attached files are confidential, proprietary, and may also be legally privileged information, and are intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient of this e-mail, please send it back to the person who sent it to you and delete the e-mail and any attached files and destroy any copies of it; you may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED] Quinnox Consultancy Services and/or any of its sister companies owns no responsibility for the views presented in the e-mail and any attached files unless the sender mentions so, with due authority of Quinnox Consultancy Services. Unauthorized reading, reproduction, publication, use, dissemination, forwarding, printing or copying of this e-mail and its attachments is prohibited. We have checked this message for any known viruses; however we decline any liability, in case of any damage caused by a non-detected virus. For more details about our company, visit http://www.Quinnox.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080116/47 00a7bf/attachment-0001.htm ------------------------------ Message: 6 Date: Wed, 16 Jan 2008 19:29:20 +0000 From: Simon Elliston Ball <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] asterisk to mysql database! To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=UTF-8; format=flowed; delsp=yes Try: http://www.voip-info.org/wiki/view/Mysql and the links thereon. simon Simon Elliston Ball [EMAIL PROTECTED] On 16 Jan 2008, at 19:11, Naveen Palani wrote: > Hello, > > Is there a possibility to connect from asterisk to mysql database > without the interface application like Ruby or PHP. > > If i can connect to mysql database from asterisk, i can update the > database for manipulations. > > Appreciate your response. > Regards, > Naveen.Palani > > > ?Quinnox, a global IT services company prides itself on its SEI-CMM > Level 5, ISO?9001:2000 assessed delivery processes and provides > solutions in areas of E-Business, ERP, Application Management > Services, and EAI to customers in BFSI, Manufacturing, Retail, > Telecom and Healthcare sector, powered by our Global Delivery > Model.? > > This e-mail and any attached files are confidential, proprietary, > and may also be legally privileged information, and are intended > solely for the use of the individual or entity to whom they are > addressed. If you are not the intended recipient of this e-mail, > please send it back to the person who sent it to you and delete the > e-mail and any attached files and destroy any copies of it; you may > call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED] > > Quinnox Consultancy Services and/or any of its sister companies owns > no responsibility for the views presented in the e-mail and any > attached files unless the sender mentions so, with due authority of > Quinnox Consultancy Services. > > Unauthorized reading, reproduction, publication, use, dissemination, > forwarding, printing or copying of this e-mail and its attachments > is prohibited. > We have checked this message for any known viruses; however we > decline any liability, in case of any damage caused by a non- > detected virus. > > For more details about our company, visit http://www.Quinnox.com > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ------------------------------ Message: 7 Date: Wed, 16 Jan 2008 15:24:04 -0500 From: Robert Moskowitz <[EMAIL PROTECTED]> Subject: [asterisk-users] Asterisk 1.4.17 and RXFAX via T38 To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="US-ASCII"; format="flowed" I was pointed to the following: http://asteriskforum.ru/viewtopic.php?t=1761 It is in Russian, which I don't speak, but it references an Asterisk patch. Is this patch in 1.4.17? Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?) Anyone work with this? ------------------------------ Message: 8 Date: Wed, 16 Jan 2008 20:39:50 -0000 From: "Chris Bagnall" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Unable to open master device '/dev/zap/ctl' To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="UTF-8" Make sure asterisk is in the "dialout" group in /etc/passwd The default gentoo ebuild of zaptel creates /dev/zap/* with group dialout, and if you're using the gentoo ebuild of asterisk, it'll run as asterisk:asterisk, so you need to make sure asterisk is a member of the dialout goup otherwise it'll never be able to access /dev/zap/* FWIW, as a fellow Gentoo user, 2006.1 is a bit dated, and you'd be well worth updating to 2007.0 if you can spare the time - it'll save you a lot of messing around with gcc versions etc. later down the line. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ------------------------------ Message: 9 Date: Wed, 16 Jan 2008 23:01:55 +0100 From: Vincent <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones? To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii On Wed, 16 Jan 2008 18:08:23 +0000 (GMT), Gordon Henderson <[EMAIL PROTECTED]> wrote: >However, you'll need to do similar things to your asterisk box & router if >it's behind NAT for IAX as you do for SIP. (You will need a static IP >address on the NAT router and port-forward 4569 to the asterisk box, just >as you'd port-forward 5060 and 10000-20000 for SIP) Am I wrong to understand that IAX only needs one port, TCP4569 by default? So I only need one port for each phone, while SIP requires at least 3 (SIP, and one RTP each way)? >And a SIP phone behind a NAT router is also solvable if it supports STUN. But not all NAT routers support STUN, ie. keeping UDP ports open so that incoming packets can make it. >I know that SIP behind NAT isn't perfect, but with care, it's very usable >and workable But unless I'm mistaken, when NAT is involved, canreinvite must be set to no, ie. all RTP packets must go through Asterisk instead of flowing from one phone to the other? Thanks guys. ------------------------------ Message: 10 Date: Wed, 16 Jan 2008 23:22:10 +0100 From: Vincent <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Can DB() use SQLite instead of BerkeleyDB? To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii On Wed, 16 Jan 2008 12:10:35 -0600, Tilghman Lesher <[EMAIL PROTECTED]> wrote: >No, it cannot. You could use func_odbc to formulate your own queries, >though. Thanks. I don't like ODBC, but if it's stable and not a pain to install/use, that could be the solution. Otherwise, there's a new solution to use MySQL: http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL ------------------------------ Message: 11 Date: Wed, 16 Jan 2008 16:26:03 -0600 From: Tilghman Lesher <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] asterisk to mysql database! To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="utf-8" On Wednesday 16 January 2008 13:29:20 Simon Elliston Ball wrote: > Simon Elliston Ball > [EMAIL PROTECTED] > > On 16 Jan 2008, at 19:11, Naveen Palani wrote: > > Hello, > > > > Is there a possibility to connect from asterisk to mysql database > > without the interface application like Ruby or PHP. > > > > If i can connect to mysql database from asterisk, i can update the > > database for manipulations. > > Try: > http://www.voip-info.org/wiki/view/Mysql > > and the links thereon. Or read configs/func_odbc.conf.sample. -- Tilghman ------------------------------ Message: 12 Date: Wed, 16 Jan 2008 22:45:59 +0000 (GMT) From: "Tim H. Panton" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=utf-8 ----- Original Message ----- From: "Vincent" <[EMAIL PROTECTED]> To: [email protected] Sent: 16 January 2008 22:01:55 o'clock (GMT) Europe/London Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones? On Wed, 16 Jan 2008 18:08:23 +0000 (GMT), Gordon Henderson <[EMAIL PROTECTED]> wrote: >However, you'll need to do similar things to your asterisk box & router if >it's behind NAT for IAX as you do for SIP. (You will need a static IP >address on the NAT router and port-forward 4569 to the asterisk box, just >as you'd port-forward 5060 and 10000-20000 for SIP) Am I wrong to understand that IAX only needs one port, TCP4569 by default? So I only need one port for each phone, while SIP requires at least 3 (SIP, and one RTP each way)? -------- That's UDP 4569. Also, depending on your configuration, you may not need to do any port forwarding at the 'client' end. Just have all your phones send registrations frequently and your natting router will do the rest. Tim. ------------------------------ Message: 13 Date: Wed, 16 Jan 2008 15:52:21 -0800 From: Steven <[EMAIL PROTECTED]> Subject: [asterisk-users] HDLC errors To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed I'm running Asterisk 1.2.26.1 svn rev 79171 on Trixbox 2.2. libpri 1.2.7 and zaptel 1.2.22.1. The hardware is a HP dl360 single cpu with a TE220B. The system load is below 0.10. I moved the server into production, with one PRI, on Friday. On that day we handled a couple thousand calls and I only saw one HDLC abort message. On Saturday half the calls and two abort messages an hour apart. On Sunday, after 1500 when there was only a couple calls, the HDLC messages went crazy. We're getting non-stop Abort messages, with Bad FCS thrown in about every tenth message. They come in bunches, with short 10-30 second breaks. Then every once and awhile there is an 30 minute break, sometimes a 3 hour break. The messages seems completely separate from system load. The system will be idle and get the messages and have no messages when I load up dozens of calls on it (using call files to complete calls) After reading the mailing list and various websites (asteriskguru.com has a couple articles), the first thing I did was look for IRQ conflicts. The module for the usb bus (no usb devices attached) was on the same IRQ. Disabling USB had no effect. zttool shows no IRQ misses. The second PRI was installed on Monday, that day with only two calls, the message came 11 times. Three times on Tuesday with no calls, then late at night I loaded it up with calls for testing (having call files call out on the second PRI to the first PRI) and no messages were generated. Again today its had a few messages with only a couple calls. I'm not sure what to try next, other than calling the telco and asking them to check their equipment. Does any one have a suggestion before I do that? Thanks. ------------------------------ Message: 14 Date: Wed, 16 Jan 2008 18:07:20 -0600 From: Russell Bryant <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] HDLC errors To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Steven wrote: > I'm not sure what to try next, other than calling the telco and asking > them to check their equipment. Does any one have a suggestion before I > do that? I have a suggestion. Have you contacted Digium technical support for assistance with resolving this issue? -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ------------------------------ Message: 15 Date: Wed, 16 Jan 2008 17:11:44 -0700 (MST) From: [EMAIL PROTECTED] Subject: [asterisk-users] AddQueueMember and Flash Operator Panel To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain;charset=iso-8859-1 Hello users! Recently I read that AgentCallbackLogin is going to be deprecated soon. Wanting to set up a few callback type queues, I set them up as suggested in queues-with-callback-members.txt. I was able to set the queues up completely this way, however, I'm trying to use Flash Operator Panel (aka AsterNIC) to monitor the agents' login status. FOP monitors their status if I call AddQueueMember with their actual interface (which, by the way, makes more sense to me than logging them in via chan_local), and it even seems to work with Local/[EMAIL PROTECTED] But if I use any context other than "default" here, FOP doesn't recognize that the agent is logged in. (The users' default context isn't even set to default, and it behaves this way even if the users' voicemail context is something else, so I am guessing that is hard-coded in FOP somewhere.) If I log them in from Local/[EMAIL PROTECTED], FOP works and the agents get the calls, but then it's just dialing them directly - there is no way to increment OUTBOUND_GROUP or check the value of GROUP_COUNT. As a result, calls are routinely sent to agents who are already on the phone, which I don't want. Obviously, the next reasonable solution would be to use some other context for the default context, and use [default] instead of [agents] for incrementing OUTBOUND_GROUP and checking GROUP_COUNT, but I'm pretty sure this would break the functionality of AsteriskGUI almost completely, and I'm trying to preserve as much of that as possible. Am I missing something? Is there a way to make all of this work together without modifying some source code? Thanks in advance! Jason Burbage [EMAIL PROTECTED] ------------------------------ Message: 16 Date: Wed, 16 Jan 2008 19:32:04 -0500 From: "Steve Totaro" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] HDLC errors To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" On Jan 16, 2008 7:07 PM, Russell Bryant <[EMAIL PROTECTED]> wrote: > Steven wrote: > > I'm not sure what to try next, other than calling the telco and asking > > them to check their equipment. Does any one have a suggestion before I > > do that? > > I have a suggestion. Have you contacted Digium technical support for > assistance > with resolving this issue? > > -- > Russell Bryant > Senior Software Engineer > Open Source Team Lead > Digium, Inc. > Excellent suggestion. Make sure you can give them SSH access and screen so you can see what they are doing. Before that, check (remake) your T1 cables and if it is punched down on a block, re-punch it. Work with your telco as well. I call that burning the candle from both ends. You said there were no errors looping port one to port two and generating calls with call files. That may indicate a telco issue. I usually open a ticket with the telco right away just in case so it can be escalated quicker if in fact it is the telco. Sometimes you have to be a jerk to these guys to get someone with half a brain to look into your problem rather than blaming CPE (the easiest way to close their ticket and get you off the phone). If Digium says it is the telco and the telco says it is your CPE (Asterisk/Digium/Server/CPE wiring) then put them together on a conference call! I am sure it won't come to that if it is truly a Digium/Asterisk issue. They will take care of it. Thanks, Steve Totaro -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080116/18 dcd050/attachment-0001.htm ------------------------------ Message: 17 Date: Wed, 16 Jan 2008 19:39:34 -0500 From: "Steve Totaro" <[EMAIL PROTECTED]> Subject: [asterisk-users] Anyone Using a Dell PowerEdge T105 in Production To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Unbeatable price for a low end Asterisk server (or any server for that matter) http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l =en&oc=bednv4k&s=bsd I wonder if anyone has any experience with this box and Digium or Sangoma hardware? Any compatibility issues? If not, I might stock up on them. Thanks, Steve Totaro -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080116/3f 9d7ae1/attachment-0001.htm ------------------------------ Message: 18 Date: Wed, 16 Jan 2008 18:42:53 -0600 From: "Ruben Zamora" <[EMAIL PROTECTED]> Subject: [asterisk-users] Problem with a channel To: <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" I have install a Server with Centos 1 TDM400: Asterisk 1.4.9, Zaptel 1.4.5 I having these problem : Zap/2-1 is busy Hangup ZAP/2-1 Everyone is busy/congested at this time (1:1/010) Autofallthrough channel "SIP/202-b7b08ab0" Status is busy. And then HANGUP. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080116/22 0b1839/attachment-0001.htm ------------------------------ Message: 19 Date: Wed, 16 Jan 2008 19:44:54 -0500 From: "Andrew Joakimsen" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] HDLC errors To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=UTF-8 Trixbox 2.2... I assume you are using the latest version. Normally I will ignore messages from trixbox users because they ask kindergarten stuff... but you seem to be knowledgeable and I'll assume you chose trixbox to make your life easier when it comes to dealing with others regarding the PBX. I also assume the PRI is delivered via some sort of HDSL terminated at an NIU ("SmartJack") Which is a box that will usually have 2 or 4 positions for line cards and 2 or 4 jacks marked "CPE1" etc.... usually at the bottom. Usually also you can look through the window at the top and see various lights. What is between the smartjack and your T1 card? What sort and length of cable? Any splices? Punchdown or patch panels? Also I'm not sure if Trixbox has this but ssh in and see if there is an application called zttool. What are the statistics it is providing? ------------------------------ Message: 20 Date: Wed, 16 Jan 2008 18:54:47 -0600 From: KodaK <[EMAIL PROTECTED]> Subject: [asterisk-users] IMAP client in asterisk not trying to contact IMAP server To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 I'm trying to test IMAP in 1.4.17 and it appears to be not working. I've compiled imap-2007 with the following on a CentOS 5 box: make slx EXTRACFLAGS="-I/usr/include/openssl -fPIC" and I've configured and compiled asterisk with the following: ./configure --with-imap=/usr/local/src/imap-2007 The compile and install went just fine, no warnings and no errors that I saw. However, when actually trying to use it, it doesn't appear that asterisk is even trying to use the local IMAP server. The local IMAP server is dovecot, with a master password configured. I've tried plain and SHA auth, but from the logs I don't even see the asterisk master user trying to connect. Here's my voicemail.conf: [general] imapserver=localhost imapfolder=Inbox ;pollmailboxes=yes ;pollfreq=30 imapflags=notls authuser=asttest expungeonhangup=yes authpassword=whatever [default] 5252 => 5252,Test,[EMAIL PROTECTED],,imapuser=5252 (I have also tried this line as: 5252 => 5252,Test,,,imapuser=5252 5252 => 5252,Test,[EMAIL PROTECTED],,imapuser=5252|imappass=pass 5252 => 5252,Test,,,imapuser=5252|imappass=pass all with and without the authuser and authpassword in the general section.) I can authenticate against the * server using 5252*asttest as the username and "whatever" as the password, which I'm lead to believe is how * will try to connect. (Also, the imap user 5252 exists and can receive mail.) Is there something else I'm missing? Is there some other place in the dial plan that I have to say "use IMAP"? Is there some way to confirm that the imap client has been compiled in? Some hidden CLI command to debug it? doing "grep -i imap /var/log/asterisk/*" gives absolutely no results. I'm almost convinced that I've got something wrong in the configuration because I tried the latest SVN and I didn't see it hit the IMAP server, but it also segfaulted so who knows. Any ideas at all? Am I missing something obvious that I'll find as soon as I press "send" and wish I hadn't sent the message? Thanks, --J(K) ------------------------------ Message: 21 Date: Wed, 16 Jan 2008 17:55:13 -0700 (MST) From: [EMAIL PROTECTED] Subject: [asterisk-users] Asterisk Now Beta 6 and CISCO IP 7910 To: "[email protected]" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain;charset=iso-8859-1 The phones are configured in the "Users" section of AsteriskGUI. The bigger problem you'll have is that you probably also need to replace/update the firmware on the 7910; by default they're configured to work with Cisco's CallManager software. Start with this link: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx Hope that helps. Good luck! Jason Burbage ------------------------------ Message: 22 Date: Wed, 16 Jan 2008 19:11:22 -0600 From: "Erik Anderson" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in Production To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 On Jan 16, 2008 6:39 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > Unbeatable price for a low end Asterisk server (or any server for that > matter) > > http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l =en&oc=bednv4k&s=bsd > > I wonder if anyone has any experience with this box and Digium or Sangoma > hardware? Any compatibility issues? If not, I might stock up on them. Wow - that *is* a great price. I don't have any of this particular box in production, but I do have 2 PowerEdge SC440s (one step up from the T105) running asterisk along with Sangoma PRI cards. They're working great. I really only have two issues with these low-end servers: 1. You can't order 'em with RAID support. I'm getting around this by using software RAID1 in linux, but I'd much prefer having a hardware RAID controller. 2. The Dell DRAC remote management cards aren't compatible with these low-end server motherboards. I've become *completely* addicted to the DRAC cards on the high-end PowerEdges, to the point that I now refuse to order a server without a DRAC card. That said, I'm sure this server would run a small/medium asterisk install just fine. -Erik ------------------------------ Message: 23 Date: Wed, 16 Jan 2008 20:28:58 -0500 From: "Steve Totaro" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in Production To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" On Jan 16, 2008 8:11 PM, Erik Anderson <[EMAIL PROTECTED]> wrote: > On Jan 16, 2008 6:39 PM, Steve Totaro <[EMAIL PROTECTED]> > wrote: > > Unbeatable price for a low end Asterisk server (or any server for that > > matter) > > > > > http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l =en&oc=bednv4k&s=bsd > > > > I wonder if anyone has any experience with this box and Digium or > Sangoma > > hardware? Any compatibility issues? If not, I might stock up on them. > > Wow - that *is* a great price. I don't have any of this particular > box in production, but I do have 2 PowerEdge SC440s (one step up from > the T105) running asterisk along with Sangoma PRI cards. They're > working great. I really only have two issues with these low-end > servers: > > 1. You can't order 'em with RAID support. I'm getting around this by > using software RAID1 in linux, but I'd much prefer having a hardware > RAID controller. > 2. The Dell DRAC remote management cards aren't compatible with these > low-end server motherboards. I've become *completely* addicted to the > DRAC cards on the high-end PowerEdges, to the point that I now refuse > to order a server without a DRAC card. > > That said, I'm sure this server would run a small/medium asterisk > install just fine. > > -Erik > You can add the raid option for $199. I think I might pickup about ten of them at this price. I can always resell them as general purpose servers or even workstations if Asterisk/Zaptel/Linux does not like the boxen. Thanks, Steve Totaro -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080116/94 021181/attachment-0001.htm ------------------------------ Message: 24 Date: Wed, 16 Jan 2008 17:34:01 -0800 From: "shadowym" <[EMAIL PROTECTED]> Subject: [asterisk-users] Asterisk on ClarkConnect To: <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Has anyone tried installing Asterisk on ClarkConnect? It looks like ClarkConnect runs on RHEL so it should work if they haven't modified it too much. It appears that ClarkConnect is working on adding Asterisk and integrating it into their GUI but until then I'd also be interested in trying to use FreePBX. Anyone? ------------------------------ Message: 25 Date: Wed, 16 Jan 2008 20:48:15 -0500 From: "Walter Willis" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Unable to open master device '/dev/zap/ctl' To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" create nodes and links /proc/zap On Jan 16, 2008 3:39 PM, Chris Bagnall <[EMAIL PROTECTED]> wrote: > Make sure asterisk is in the "dialout" group in /etc/passwd > > The default gentoo ebuild of zaptel creates /dev/zap/* with group dialout, > and if you're using the gentoo ebuild of asterisk, it'll run as > asterisk:asterisk, so you need to make sure asterisk is a member of the > dialout goup otherwise it'll never be able to access /dev/zap/* > > FWIW, as a fellow Gentoo user, 2006.1 is a bit dated, and you'd be well > worth updating to 2007.0 if you can spare the time - it'll save you a lot > of messing around with gcc versions etc. later down the line. > > Regards, > > Chris > -- > C.M. Bagnall, Director, Minotaur I.T. Limited > For full contact details visit http://www.minotaur.it > This email is made from 100% recycled electrons > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080116/77 47e7e2/attachment-0001.htm ------------------------------ Message: 26 Date: Wed, 16 Jan 2008 19:54:13 -0600 From: "Erik Anderson" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in Production To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 On Jan 16, 2008 7:28 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > > You can add the raid option for $199. I think I might pickup about ten of > them at this price. I can always resell them as general purpose servers or > even workstations if Asterisk/Zaptel/Linux does not like the boxen. Ahh - nice. That wasn't an option when I ordered the SC440. -erik ------------------------------ _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 42, Issue 58 ********************************************** ******************************************************************** This email and any attachments are confidential to the intended recipient and may also be privileged. If you are not the intended recipient please delete it from your system and notify the sender. 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