[asterisk-users] Question about 7940s and call forwarding
Hello, I need some advice on the following problem I'm trying to solve: At the office we are using 7940s as our phones, connected to an asterisk box via SIP. Pretty standard setup, nothing fancy. Everyone has an extension that comes out as a single line button on the phones, with the second line unused at this point. Certain things on our phone system are set to ring all the phones in the office (support queue fills up, etc.). I just simply do something like this: Dial(SIP/123SIP/456SIP/789) which works just fine, except if someone wants to use the phone's built-in call forwarding to send their calls to their house or cell phone. Then, any call that rings all the phones gets picked up and forwarded by that person's phone and they end up getting all the calls. Originally I had set up everyone as agents and let them forward their extensions anywhere, but the agent callback stuff makes things really unstable, plus it's cumbersome to use compared to using the phone. The solution I'm considering is to make the ring all calls ring to the second line button on all the phones and disabling call forwarding in sip.conf for those lines. I'm just not sure how a phone set to forward calls will react...will it properly not answer the calls on this line, or will it try to forward them and fail and cause the calls to get dropped? Will call transfers still work for calls that came in on the second line? Also if there's a better way of doing what I want to do I'm all ears. :) -- Joshua M Thompson [EMAIL PROTECTED] ___ --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] Weird DTMF issue
Ok, this one has me stumped. This setup was working fine Friday and now today it's just stopped working. Details: Dell 2850 running Asterisk 1.2.4. Phones are SIP phones (Cisco 7940s). Timing is done via a WCTDM card (also tried ztdummy.) All traffic in and out of this box to the PSTN is via IAX2. At some point between Friday and today DTMF stopped working right. Specifically, when you call our main # and are at the IVR, only the first digit you dial is recognized. For example if I try to dial 81 this is all I get for debugging: Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: DTMFSubclass: 8 Timestamp: 02123ms SCall: 00020 DCall: 9 [1.2.3.4:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 02123ms SCall: 9 DCall: 00020 [1.2.3.4:4569] As soon as the 8 is received the Background app stops playing as you'd expect, but it stops recognizing any more digits, and eventually times out and errors out with an invalid extension '8'. Even worse, if you try to dial anybody's direct extensions (2xx) now you end up in the support queue after it times out since the queue is option 2 (yeah I know that's a stupid IVR design, but I had to mimic the old PBX I didn't set up.) I've tried this through two different call paths, one through the PSTN and one direct from my house asterisk system (SIP/IAX2 end-to-end). It behaves the same both ways. The strange part is, while the invalid extension message is being played by Playback() all the digits I hit *are* recognized, as they show up in the iax2 debug output. It's only in the Background() app that this seems to be a problem. Any suggestions would be greatly appreciated. This has our IVR totally busted and I've tried everything I can think of so far. -- Joshua M Thompson [EMAIL PROTECTED] ___ --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
Re: [Asterisk-Users] Weird DTMF issue
On Mon, 2006-02-27 at 15:50 -0600, Rich Adamson wrote: Ok, this one has me stumped. This setup was working fine Friday and now today it's just stopped working. Details: Dell 2850 running Asterisk 1.2.4. Phones are SIP phones (Cisco 7940s). Timing is done via a WCTDM card (also tried ztdummy.) All traffic in and out of this box to the PSTN is via IAX2. At some point between Friday and today DTMF stopped working right. Specifically, when you call our main # and are at the IVR, only the first digit you dial is recognized. For example if I try to dial 81 this is all I get for debugging: Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: DTMFSubclass: 8 Timestamp: 02123ms SCall: 00020 DCall: 9 [1.2.3.4:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 02123ms SCall: 9 DCall: 00020 [1.2.3.4:4569] As soon as the 8 is received the Background app stops playing as you'd expect, but it stops recognizing any more digits, and eventually times out and errors out with an invalid extension '8'. Even worse, if you try to dial anybody's direct extensions (2xx) now you end up in the support queue after it times out since the queue is option 2 (yeah I know that's a stupid IVR design, but I had to mimic the old PBX I didn't set up.) I've tried this through two different call paths, one through the PSTN and one direct from my house asterisk system (SIP/IAX2 end-to-end). It behaves the same both ways. The strange part is, while the invalid extension message is being played by Playback() all the digits I hit *are* recognized, as they show up in the iax2 debug output. It's only in the Background() app that this seems to be a problem. Any suggestions would be greatly appreciated. This has our IVR totally busted and I've tried everything I can think of so far. It would have been helpfull if you would have posted the few dialplan entries associated with starting the ivr. The following works fine for me for incoming iax2 analog pstn calls: Here's the IVR. As I said this was working at least through Friday and was in service for quite some time with no changes: exten = s,1,Set(TIMEOUT(digit)=5) exten = s,2,Set(TIMEOUT(response)=20) exten = s,3,Answer exten = s,4,Wait(1) exten = s,5,Background(s24-intro) exten = 0,1,Playback(transfer) exten = 0,2,Set(CALLERID(name)=Operator Line) exten = 0,3,Dial(SIP/221SIP/222SIP/223SIP/224SIP/225SIP/227SIP/230SIP/231SIP/233SIP/240SIP/241SIP/242) exten = 0,4,Hangup exten = 1,1,Playback(transfer) exten = 1,2,Set(CALLERID(name)=Sales Queue) exten = 1,3,Queue(sales) exten = 1,4,Hangup exten = 81,1,AgentCallbackLogin(||@outgoing-voice) exten = 81,2,Hangup exten = 83,1,AgentCallbackLogin(||@outgoing-voice) exten = 83,2,Hangup exten = #,1,Directory(s24,extensions) exten = i,1,Playback(pbx-invalid) exten = i,n,Goto(s,5) exten = t,1,Hangup include = extensions include = s24-daytime|8:00-16:59|mon-fri|*|* include = s24-evening|17:00-17:59|mon-fri|*|* include = s24-nighttime|0:00-7:59|mon-fri|*|* include = s24-nighttime|18:00-23:59|mon-fri|*|* include = s24-nighttime|*|sat|*|* include = s24-nighttime|*|sun|*|* (the extensions context, obviously, contains the individual extensions which are 2XX. There is a support queue at option 2 that's routed differently by time of day and it's in the daytime/evening/nighttime contexts.) The problems seems to be in Background(). If I add a WaitExten(10) as s,6 and wait for the entire menu to play through I can dial an extension during the silence. But if I try it during the menu it recognizes the first digit and then times out while ignoring subsequent digits. I just can't understand why it suddenly stopped working. Nothing was changed over the weekend. -- Joshua M Thompson [EMAIL PROTECTED] ___ --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
Re: [Asterisk-Users] Weird DTMF issue
On Mon, 2006-02-27 at 17:49 -0600, Rich Adamson wrote: Well... there are only two choices: 1) something did change, or, 2) a bug has surfaced. What version of code are you running? 1.2.4 now. It was on 1.2.3 but upgrading asterisk and zaptel was the first thing I tried when we noticed the problem this morning. I assume you've tried stopping and restarting asterisk, right? Yep tried that a couple times now. Also thought maybe the TDM400 card was acting up so I switched to ztdummy, since we're only using the card for timing purposes. I'm about to try rebooting the entire machine now that it's after hours. There's nothing suspicious in the logs or dmesg output but rebooting is about the only thing I haven't tried yet. -- Joshua M Thompson [EMAIL PROTECTED] ___ --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
Re: [Asterisk-Users] 7940 paperweight
On Fri, 2005-11-11 at 16:02 -0700, Kris Edwards wrote: Here is a tcpdump (mac changed): 15:48:31.501856 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:35.501998 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:39.502162 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:43.502293 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:47.504194 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:48:51.502542 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:48:55.502685 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:48:59.502815 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:49:03.502961 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:49:07.544093 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 15:49:08.033496 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 15:49:09.033501 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 15:49:11.033569 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 This looks strange to me...your server appears to be completely ignoring the tftp requests from the phone. It's not even responding with a connection refused. I know you said you didn't have a firewall between the phone and the server but this is exactly the behavior I'd expect from a firewall with a default policy of drop. If it's a RedHat box of some sort (Fedora, RHEL, Centos) you might want to make sure the default firewall is turned off by running service iptables stop as root. -- Joshua M Thompson [EMAIL PROTECTED] ___ --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] Status of linux 2.6 support
On Fri, 2004-12-03 at 08:39 -0500, Clint Guillot wrote: Any ideas on why the T1 cards won't come out of red alarm, but work fine on the same machine booted to 2.4? It's quite possible you have an interrupt-related problem in 2.6 but not in 2.4, since 2.6 tries to fully utilize ACPI now. You might want to try booting with acpi=off and see if that fixes the situation. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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 1.0 RPMS RH73 and RH9
If nobody else has them I can rebuild Greg's RPMs on one of my FC2 machines and post them. It'll just have to wait until I'm back at the hotel later tonight. On Thu, 2004-09-23 at 13:49 -0700, Chad Brown wrote: Is anyone working on a Fedora Core 2 RPM? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: Thursday, September 23, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9 Hello, Straight from the floor of Astricon 2004, I am happy to release my updated Asterisk 1.0 RPMS for RedHat 7.3 and RedHat 9.0 platform. Current Release --- asterisk-1.0-0 libpri-1.0-0 zaptel-1.0-0 kernel-module-zaptel-1.0-0 RedHat 7.3 -- ftp://ftp.nacs.net/asterisk/rh73/RPMS/ ftp://ftp.nacs.net/asterisk/rh73/SRPMS/ RedHat 9.0 -- ftp://ftp.nacs.net/asterisk/rh9/RPMS/ ftp://ftp.nacs.net/asterisk/rh9/SRPMS/ Changelog - * Thu Sep 23 2004 Gregory Boehnlein [EMAIL PROTECTED] - Updated to version 1.0 - Drank beer at Astricon * Thu Aug 12 2004 Gregory Boehnlein [EMAIL PROTECTED] - Updated to version 1.0-RC2 - Replaced parking.conf with features.conf * Sat Jul 17 2004 Gregory Boehnlein [EMAIL PROTECTED] - Updated to version 1.0-RC1 * Thu May 27 2004 Gregory Boehnlein [EMAIL PROTECTED] - Updated to version 0.9.0 * Wed Feb 04 2004 Gregory Boehnlein [EMAIL PROTECTED] - Updated to version 0.7.2 * Sat Jan 31 2004 Gregory Boehnlein [EMAIL PROTECTED] - Updated development environment to ensure proper build consistency for chan_zap - Added post-install chkconfig to auto-start asterisk on boot - First really useable release. Yay! * Mon Jan 26 2004 Gregory Boehnlein [EMAIL PROTECTED] - Updated changelog entry to enable build on Fedora Core 1 [EMAIL PROTECTED] - Made the decsision to use Dist Specific version numbers (_fc1,_rh9,_rh8,_rh73) * Sat Jan 24 2004 Gregory Boehnlein [EMAIL PROTECTED] - added doc macros - added config macros - updated install stanza to correct symlink issue - updated patch0 to include changes to Makefile - added /etc/rc.d/init.d/asterisk - added export LD_ASSUME_KERNEL=2.4.1 for RH9 - asterisk.spec now builds cleanly on RH73 and RH9 * Wed Jan 21 2004 Gregory J. Boehnlein [EMAIL PROTECTED] - Initial .spec file created. Most likely buggered. Badly needs help. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Astricon meets?
On Wed, 2004-09-22 at 00:19 -0400, Joshua M. Thompson wrote: More importantly, who else is at the other here...the Marriot Marquis. I was one of the unlucky ones who didn't get a room at the Century Center. Oh and on a related note...I'm here, I'm bored, and I'm hungry. If anyone else is interested I wouldn't mind hitting the town looking for a bite to eat and/or maybe a friendly bar. ;-) -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Astricon meets?
As I understand it we can either stay at the Marquis for the duration or try again tomorrow to move to the CC. I'm probably not going to go through the hassle of packing back up and moving over to the CC, especially since I already paid for the STSN net access for the duration of my stay here. :) I drove down here so I have transportation between the hotels, once I get over there tomorrow to pick up my car again. On Tue, 2004-09-21 at 22:28 -0600, Brandon Patterson (peering) wrote: What are they going to do about that anyway? - Original Message - From: Joshua M. Thompson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, September 21, 2004 10:19 PM Subject: Re: [Asterisk-Users] Astricon meets? More importantly, who else is at the other here...the Marriot Marquis. I was one of the unlucky ones who didn't get a room at the Century Center. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] MySQL on another host?
Oliver Breidenbach wrote: Hi there, what do I need to take into consideration if I want Asterisk to talk to a MySQL database on a different host to store CDR records? The cdr_addon_mysql module does not want to load and Asterisk claims that it cannot open shared object. It compiled fine, however. It sounds like you don't have the MySQL client libraries properly installed, although you seem to have the headers since the module compiled. If your mysql client libs are not in /lib, /usr/lib or /usr/local/lib then you'll probably have to add whatever directory they are in to /etc/ld.so.conf and rerun ldconfig. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Commercial CID spoofing system
On Thu, 2004-09-02 at 20:54 +0100, Kevin Walsh wrote: William Suffill [EMAIL PROTECTED] lazily top-posted: star38.com .25 connection .07-.13 per min What a bargin Was there a point to that, or was that just spam? You didn't quote any context for your statement, so I have no idea what you are referring to or answering. Star38.com is a caller ID spoofing service that was recently announced. You subscribe to the service, and then when you want to make a call you fill out a web form with your phone #, the number you want to call, and the CallerID you want to appear on your call. The system then calls you and the victim and bridges the calls together. Supposedly they're going to only sell this service to what they consider legitimate customers (licensed private investigators, etc.) Looks like an Asterisk box and a simple CGI script to me. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Commercial CID spoofing system
On Thu, 2004-09-02 at 22:44 +0200, Stefan de Konink wrote: Joshua M. Thompson wrote: Looks like an Asterisk box and a simple CGI script to me. Is this possible out there without a SS7 gateway? Or do you need just a friendly channel supplyer that allows you to send any callerids thru their switches? It can be done with a PRI as long as you just have a friendly provider. It's not hard to find one at least in the US. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Cisco 79xx series IP phones
On Fri, 2004-08-13 at 11:31 -0500, Shawn Parker wrote: Does anyone have any knowledge or experience to give me dealing with Cisco 7902G and 7905G IP phones and getting them to work on a lan with Asterisk when *not* using other Cisco hardware? The sales guy is just trying to sell you more Cisco hardware. I have 7910s, 7940s and 7960s at my house all running just fine on a Linksys 24-port fast ethernet switch with zero problems. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] tdm400p lockups
On Fri, 2004-08-06 at 11:19 -0400, Tim Sailer wrote: I have a system that has a TDM400 with 3 FXO modules. Recently, the system started not seeing the ports. No answer, no lines available for oubound dial, and only a reboot of the machine will bring it back. Any ideas to try? You too? Mine (two FXO/two FXS) started doing that, and I had to keep unloading and reloading the driver to bring it back. Then about a week ago it just stopped seeing modules period, no matter what I do. I'm about to test it on another machine to make sure it's really dead and then RMA it. Maybe a bad batch of cards? When did you buy yours? I got mine mid-June, right after FXO modules became (briefly) available again. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] tdm400p lockups
On Fri, 2004-08-06 at 10:16 +, Ryan Courtnage wrote: Are you using a newer zaptel build? If so, make sure you didn't get caught by bug #2218. wcfxs had an error introduced this week, which is now resolved in HEAD. It's possible, although I did try using zaptel sources from another server of mine that was a couple weeks out of date and it didn't seem to help. But I will try again tonight when I get home; I plan to test it on another machine first just to make sure my motherboard or power supply hasn't gone insane. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] tdm400p lockups
On Fri, 2004-08-06 at 20:39 -0400, Greg Boehnlein wrote: I'd suggest slapping the TDM400P in an Apple ][ GS and trying the new Zaptel Drivers for METAL. ;) Hmm that will be perfect for my new multiline Future Vision BBS ;-) -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Cisco SIP Phone 7960 DTMF Problem
On Wed, 2004-08-04 at 04:35, Miroslav Nachev wrote: Hi, When we use BudgeTone where the DTMF is set to via RTP (RFC2833) all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to remote pick-up the call through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Is your cisco dial plan file set up to allow you to dial *8#? -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Echo problem update - POSSIBLE SOLUTION
On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote: No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes? My guess would be interrupt and/or PCI latency. Echo is produced by delays in the audio path so if some motherboards are adding delays it's going to make the echo worse. Fiddling with PCI bus settings both in the BIOS and from Linux (using the pci tools) may help in some cases. The unfortunate part about this is that there are SO many variables that can influence latency that you can't really tell if a motherboard is going to work or not until you try it. Even two MBs with the same CPUs and the same north/south bridges could produce different results. Probably the best we can hope for right now is to start building a whitelist of known good motherboards for people to reference when building Asterisk systems. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] CISCO 7960G FIRMWARE
On Wed, 2004-07-14 at 17:19 -0400, Kanuri, Seshu wrote: Hi All, CISCO 7960G is may not be a good decision. Without Call Manager, this is garbage. A friend of mine bought this a couple of days ago and had to return quickly, to get the refund. I run a couple several 7960G and 7940G phones at home using SIP (no Call Manager) and they are the best phones I've ever owned. Was your friend just unable to find SIP firmware for them? -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Help Needed in configuring Cisco 7940
On Tue, 2004-07-13 at 14:43, Shaun Ewing wrote: Note: Cisco software images are only available from Cisco's web site and are protected by copyright laws. Access to their web site requires an account be established. The easiest way to do that is to purchase a Maintenance Agreement from Cisco for approximately $8 per year (US). Then the trick is just finding someone who will sell it to you. After a couple arguments with Cisco resellers I gave up. I bet Cisco could make some easy money by whipping up a PHP or Perl script that would let you sign up for IP phone SmartNet contracts via a simple web form (enter your name, billing address, CC#, phone model and serial #). -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Cisco 7960G and *
On Fri, 2004-07-02 at 10:45, Matt Davies | MattDavies.Net wrote: I have been doing so much reading on phones lately that I have completely lost track of some things. I seem to remember that there was one series of Cisco IP phones that required Cisco's call manager. Does anyone know if the 7960 will work with Asterisk or does it require call manager? 7940 and 7960 work fine with SIP. The 7905G (but *not* the regular 7905) and 7912G should work as well. I think the ones that *require* Call Manager are: 7902 7910 (too bad, it's a nice little inexpensive phone) 7920 7970 (a VERY nice phone, wish it did SIP) -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Re: 7960 straight through?
On Fri, 2004-06-18 at 13:03, Randy Bush wrote:
if i go off hook and dial 666 from an internal sipura spa-x000
(at extn 141), it rings straight through to extn 666.
using the same dialplan, from a cisco 7960 with 7.1 sip code
(at extn 142), i have to
go off hook
hit NewCall
punch 142 (or any valid extn in the dialplan) the problem ***
hit Dial
then dial 666
sip.conf for crisco
[fiji]
callerid=crisco 142
type=friend
host=dynamic
port=5060
secret=pfui
qualify=1000
dtmfmode=rfc2833
canreinvite=yes
context=in-internal
extensions.conf
[in-internal]
exten = s,1,Answer
exten = 141,1,GoTo(int-extns,s,1) ; spa-x000
exten = 142,1,GoTo(int-extns,s,1) ; 7960
[in-extns]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,PlayTones(dial)
exten = 141,1,Macro(dial-extension,marais)
exten = 142,1,Macro(dial-extension,fiji)
exten = 666,1,Macro(dial-extension,downthere)
The reason you're getting this behavior from the Cisco is that you have
assigned it to the in-internal context. That context has no way out
other than to dial a valid extension. Once you do that it transfers to
the in-extns context, where 666 is valid. I bet yourother phone is set
to be in the in-extns context so it doesn't need to do this to dial
out.
Just out of curiosity why do you have this strange setup? I usually use
a setup something like this:
[extensions]
exten = 101,1,Macro(vmextension,101,${EXTEN101})
exten = 102,1,Macro(vmextension,102,${EXTEN102})
[pstn]
exten = _NX,1,Macro(route,${EXTEN})
[applications]
exten = *98,1,VoicemailMain(${CALLERIDNUM})
[speeddials]
exten = #01,1,Macro(route,2345678901)
[internal]
include = extensions
include = applications
include = speeddials
include = pstn
(where the 'route' macro is a macro that looks up the NPA/NXX via dbodbc
and routes local calls to my analog trunks and long distance calls to my
VoIP trunk)
And then all my cisco phones are set to be in the internal context and
they can dial any internal extension as 1XX or dial a plain ten digit
PSTN number. There won't be a conflict because my dialplan uses strict
10D dialing (no 1+number) so anything beginning with 1 cannot be a valid
PSTN number. So my dialplan.xml is set to allow 1XX to dial immediately.
If you need more help with your dialing plan email me off list. I have
four Cisco phones in my house (1 7960G and three 7940Gs) and they're all
working just fine without problems using SIP with firmware version 7.1.
--
Joshua M. Thompson [EMAIL PROTECTED]
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Re: [Asterisk-Users] Fedora Core 2 and Kernel 2.6
On Thu, 2004-05-20 at 05:12, WipeOut wrote: When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with.. make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Anyone got ant ideas? You'll need to configure the source tree before zaptel will compile. The config files are in /usr/src/linux-2.6/configs...copy the one that matches what you're running to /usr/src/linux-2.6/.config and then run make oldconfig. Zaptel should compile after that. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Fedora Core 2 and Kernel 2.6
On Thu, 2004-05-20 at 12:04, WipeOut wrote: Thanks for the try but its didn't work.. Got exactly the same result.. Apparently the FC2 2.6.5 kernel has another issue, one that I didn't start seeing until 2.6.6 (I build my own kernel RPMs.) There are a few files that are auto generated by the Makefile. The complete directions to set up your source tree are thus: cp configs/config-for-my-kernel .config make oldconfig make include/asm make include/linux/version.h make SUBDIRS=scripts A pain in the butt but at least you only have to do this once after installing a new kernel-source RPM. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Fedora Core 2 and Kernel 2.6
On Thu, 2004-05-20 at 13:23, WipeOut wrote: Well done Joshua!!.. I have no idea what all that just did but it looks like Zaptel has built.. I won't be able to test the drivers for a while with and actual card but at least i can now try build Libpri and Asterisk.. You should be fine now...I just upgraded my home Asterisk box to FC2 last night (which is how I knew what to do) and it's working fine. -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] What's in ${EXTEN} ? Why does voicemail prompt for an extension?
On Fri, 2004-05-14 at 14:47, Paul Mahler wrote:
Why does voicemail prompt me for an extension instead of just asking my
password?
[voice-mail]
exten = 99,1,VoicemailMain([EMAIL PROTECTED])
exten = 99,2,Hangup
The ${EXTEN} variable is the extension number, which in this case will
always be 99. You will want to use ${CALLERIDNUM} here (or some variant
thereof.)
--
Joshua M. Thompson [EMAIL PROTECTED]
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[Asterisk-Users] US source for compatible ISDN cards?
Is there a source out there for getting Linux/Asterisk compatible ISDN PCI cards in the US, *besides* the overly pricey Eicon Diva Server? Linux Central doesn't seem to carry the NETspider-U anymore so I'm looking for either that or an inexpensive passive ISDN card to play with. I've scoured ebay and found nothing so far. :( If someone out there has a card or two and wants to sell it I'll happily PayPal you the money plus shipping. I have an NT-1 here so S/T interface is fine; all I require is that it be PCI since I don't have ISA slots. I *do* have a TigerJet card here with a U interface, but the backend chip is a Siemens IECQ instead of the ISAC so the i4l driver won't drive it. I've been meaning to do the work necessary to get it working but just haven't had the time. -- Joshua M. Thompson [EMAIL PROTECTED] Planet Jurai ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 4-port T1 cards
On Wed, 2003-05-28 at 14:30, Steven Critchfield wrote: Don't use Mysql. if you ever have had to deal with it in a production environment that works it over, you will know that as it reaches it's limits, it starts a death spiral that is very difficult to recover from. For our software on a dual P3 866 with a gig of ram, the limit was around 1.5 queries a second fairly mixed update, inserts, and selects. Total file size of the database was under 200meg, and was fully cached so even though we had hardware raid 5 across 4 10K rpm ultra160 drives, it shouldn't have mattered for the selects. What table type? The default MyISAM tables don't support row-level locking and thus are horrible if you do a lot of inserts or updates. InnoDB tables however are much, much better. Hell SlashDot runs on MySQL with InnoDB. :) -- Joshua M. Thompson [EMAIL PROTECTED] Planet Jurai ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
