Re: [asterisk-users] [Dahdi 2.3.0.1] Does it need all those modules?
I just went through a Dahdi rebuild, and I seem to recall a message that all modules will be loaded until you set up the dahdi configuration files. regards Scott On 7/9/2010 11:41 AM, Gilles wrote: Hello To use Dahdi + Asterisk with a PCI card with a single FXO port, I just... 1. compiled and installed Dahdi 2. edited /etc/modprobe.d/dahdi.blacklist.conf to blacklist netjet and unblacklist wctdm: == # cat /etc/modprobe.d/dahdi.blacklist.conf blacklist wct4xxp blacklist wcte12xp blacklist wct1xxp blacklist wcte11xp blacklist wctdm24xxp blacklist wcfxo #blacklist wctdm blacklist wctc4xxp blacklist wcb4xxp blacklist netjet == 3. rebooted, and checked that netjet was gone and wctdm was in: == # lsmod | grep -i wc wctc4xxp 32414 0 dahdi_transcode 5751 1 wctc4xxp wcb4xxp33905 0 wcfxo 8968 0 wctdm24xxp116684 0 wcte11xp 22995 0 wct1xxp12971 0 wcte12xp 26308 0 dahdi_voicebus 39947 2 wctdm24xxp,wcte12xp wct4xxp 230713 0 wctdm 35677 0 dahdi 197809 11 xpp,dahdi_transcode,wcb4xxp,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp,wctdm crc_ccitt 1339 3 wctdm24xxp,dahdi,hisax == Does Dahdi really need all those modules, or is there another configuration file that I missed to disable unneeded modules? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conditional includes in iax.conf
Hello- For maintenance purposes, if possible I'd like to use the same iax.conf file in several different asterisk systems. However, on one of the systems only, I would like to include an IAX register command to another external system. Within iax.conf or other configuration files (other than extensions.conf), is there a way of determining what system I'm running on, and include a particular configuration item conditionally? I guess what I'm asking is there a way to conditionally include lines in a configuration depending on the value of some linux environment variable? thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conditional includes in iax.conf
On 7/7/2010 11:25 AM, Danny Nicholas wrote: -- Rather than trying to determine what system you are on, just make the included file be empty on all except the desired server. OK, thanks. I thought I might have to do it that way, which is slightly less desirable, as it makes the systems different from each other. cheers Scott -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conditional includes in iax.conf
On 7/7/2010 11:52 AM, Kevin P. Fleming wrote: On 07/07/2010 01:46 PM, Scott Stingel wrote: On 7/7/2010 11:25 AM, Danny Nicholas wrote: -- Rather than trying to determine what system you are on, just make the included file be empty on all except the desired server. OK, thanks. I thought I might have to do it that way, which is slightly less desirable, as it makes the systems different from each other. You could also enable 'execincludes' in asterisk.conf, then use #exec to execute a small script (even just a shell script) that outputs the desired iax.conf content for the server it is running on. That's much easier and more effective than trying to put conditional logic and other programming constructs into the configuration file reader. Ok, thanks Kevin. Something I haven't used before but will look into! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warning spamming for any unsynchronized ISDN port with dahdi-2.3.0.1
On 6/30/2010 3:56 PM, Alex Villacís Lasso wrote: whenever an ISDN port is in RED alarm (unsynchronized), we get a stream of warnings in /var/log/asterisk/full that look like this: [Jun 30 17:38:41] WARNING[9637] chan_dahdi.c: No D-channels available! Using Primary channel 78 as D-channel anyway! [Jun 30 17:38:41] WARNING[9638] chan_dahdi.c: No D-channels available! .. question I have is this: is this warning message something to be expected from ports with RED alarms? Or is this message a symptom of a deeper misconfiguration? Alex- On my system (D410P) the above message appears when EITHER: (a) A span is configured in dahdi-channels.conf (or chan_dahdi.conf), but nothing is plugged into it OR (b) A span is configured in dahdi-channels.conf (or chan_dahdi.conf), an E1 is plugged in, BUT signalling type is incorrectly configured (pri_cpe vs. pri_net) I agree with the other person, that a single Red Alarm message would be preferable rather than have the above message repeat forever if nothing is plugged in. You can disable it if the lines are inactive by commenting out the configuration information in dahdi-channels.conf (or chan_dahdi.conf depending on your setup) Scott -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?
On 6/22/2010 4:26 AM, A J Stiles wrote: Is anybody else using the following combination: * a TE410P card (wct4xxp driver) * a BT ISDN connection * DAHDI 2.3.0.1 * Asterisk 1.6.2.9 I'm trying to configure a new box to replace a legacy system (same hardware; some old version of Asterisk with Zaptel; works lovely but hopelessly out-of-date) and not having much joy. Specifically, I couldn't get it to see a D-channel on channel 16 of span 1. And without a D-channel, there is no way I'm going to be able to get a call in or out. This could well be because the syntax of modern /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf is slightly incompatible with the old zapata.conf and zaptel.conf files. So I guess the first question should be, has anybody else managed to make this combination work? (I'm new here and I may have missed some important information, so please ask.) Hi- I've been going through the same upgrade process recently, and had the same error (shown in your other message). I had forgotten that the equipment I was plugged in to was CPE, so I had to change my new setting for that span to NET rather than CPE. I notice in your old zapata files that you had CPE for two spans and NET for the other two, and your dahdi_chan setup is set up the same. But I'm thinking perhaps during testing you plugged a CPE on your new setup to a CPE on the other, which would produce the symptoms you see. -Scott -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI: Inbound BRI call, DDI not presented
On 6/22/2010 2:03 AM, Tzafrir Cohen wrote: On Mon, Jun 21, 2010 at 09:08:02AM -0700, Scott Stingel wrote: Hello- I have a system with one D410P and one B200P (both OpenVox). All is well with the D410P, inbound and outbound, and I can initiate calls on the B200P BRI span, but there may be something wrong with my inbound BRI setup: there is no indication of an inbound call when I dial in to it from the PSTN. When I run pri intense debug and make a call to the BRI span, I can see a message containing the DDI that I'm dialing, in this case 336027 (BT supplies only the last 6 digits of a delivered number). See debug output below... Is there anything you see in the dialplan trace itself? Also, 'intense debug' shows a lot of noise of ISDN layer 2 (Q.921). But that is normally not interesting. Do you see anything on a simple 'pri debug span 1' (only layer 3 debug)? Have I neglected to set up some needed parameter? This all worked on older boards when using bristuff, but now I want to use dahdi. My client is in the UK, connected to BT, and I have specified euroisdn as the switch type. many thanks - (snippet during inbound call to 336027) Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 008 P/F: 1 0 bytes of data -- ACKing all packets from 8 to (but not including) 8 -- Stopping T200 timer -- Starting T203 timer Shouldn't an RR be sent back? Handling message for SAPI/TEI=0/0 TEI: 0 State 7 V(S) 8 V(A) 8 V(R) 8 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0 T200 0, N200 3, T203 1 [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21 a3 70 07 81 33 33 36 30 32 37 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 30 bytes of data Handling message for SAPI/TEI=0/127 TEI: 0 State 7 V(S) 8 V(A) 8 V(R) 8 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0 T200 0, N200 3, T203 1 [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21 a3 70 07 81 33 33 36 30 32 37 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 30 bytes of data Handling message for SAPI/TEI=0/127 - Thanks, will try the less intense debug. I thought it was interesting however that the incoming DDI was in the message, but not showing up in the dialplan trace.. -Scott -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FIXED: DAHDI: Inbound BRI call, DDI not presented
Thanks to the OpenVox engineer for picking this up: I had bri_cpe for my signaling type, should be bri_cpe_ptmp. The BRI circuit on the B200P works fine now in both directions. -Scott On 6/22/2010 7:58 AM, Scott Stingel wrote: On 6/22/2010 2:03 AM, Tzafrir Cohen wrote: On Mon, Jun 21, 2010 at 09:08:02AM -0700, Scott Stingel wrote: Hello- I have a system with one D410P and one B200P (both OpenVox). All is well with the D410P, inbound and outbound, and I can initiate calls on the B200P BRI span, but there may be something wrong with my inbound BRI setup: there is no indication of an inbound call when I dial in to it from the PSTN. When I run pri intense debug and make a call to the BRI span, I can see a message containing the DDI that I'm dialing, in this case 336027 (BT supplies only the last 6 digits of a delivered number). See debug output below... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?
On 6/22/2010 9:44 AM, A J Stiles wrote: On Tuesday 22 Jun 2010, Scott Stingel wrote: Hi- I've been going through the same upgrade process recently, and had the same error (shown in your other message). I had forgotten that the equipment I was plugged in to was CPE, so I had to change my new setting for that span to NET rather than CPE. I notice in your old zapata files that you had CPE for two spans and NET for the other two, and your dahdi_chan setup is set up the same. But I'm thinking perhaps during testing you plugged a CPE on your new setup to a CPE on the other, which would produce the symptoms you see. On the current machine, spans 1 and 2 are the ISDN exchange lines (they go to the box on the wall labelled NTE2D); span 3 is connected to an Eicon Diva server card for fax sending (but that's for another day .); and span 4 is available to use as though it was another exchange line (used to be used for something once). I'm not certain that span 2 actually does anything; it may have been turned off as a money-saving measure. But the cable is still plugged in anyway. I unplugged the cables from spans 1 and 2 of the old machine, and transferred them to the new machine, leaving 3 and 4 alone for the time being. Next live testing I'll have to do tonight, once nobody else needs the phones. Yes, it sounds like you've configured it correctly, ie the same as the old machine, but just for fun you might try pri_net on one of the spans, stop and start the dahdi service and asterisk and see what happens! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?
On 6/22/2010 2:15 PM, A J Stiles wrote: [Jun 22 21:34:18] WARNING[5651]: chan_dahdi.c:4160 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! AJ- On my system (D410P) the above message appears when EITHER: (a) A span is configured in dahdi-channels.conf (or chan_dahdi.conf in your case), but nothing is plugged into it OR (b) A span is configured in dahdi-channels.conf (or chan_dahdi.conf in your case), an E1 is plugged in, BUT signalling type is incorrectly configured (pri_cpe vs. pri_net) Also, you should be able to leave everything configured in /etc/dahdi/system.conf, whether or not anything is plugged into it, so set this one up and leave it alone. Finally, you might consider using dahdi_tool to see if you are getting a Red alarm or not. Note that this works independently of asterisk, but also note further that this tool will indicate OK even if the signalling type is backwards - basically it just means that another E1 is plugged into it. -Scott -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI: Inbound BRI call, DDI not presented
Hello- I have a system with one D410P and one B200P (both OpenVox). All is well with the D410P, inbound and outbound, and I can initiate calls on the B200P BRI span, but there may be something wrong with my inbound BRI setup: there is no indication of an inbound call when I dial in to it from the PSTN. When I run pri intense debug and make a call to the BRI span, I can see a message containing the DDI that I'm dialing, in this case 336027 (BT supplies only the last 6 digits of a delivered number). See debug output below... Have I neglected to set up some needed parameter? This all worked on older boards when using bristuff, but now I want to use dahdi. My client is in the UK, connected to BT, and I have specified euroisdn as the switch type. many thanks - (snippet during inbound call to 336027) Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 008 P/F: 1 0 bytes of data -- ACKing all packets from 8 to (but not including) 8 -- Stopping T200 timer -- Starting T203 timer Handling message for SAPI/TEI=0/0 TEI: 0 State 7 V(S) 8 V(A) 8 V(R) 8 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0 T200 0, N200 3, T203 1 [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21 a3 70 07 81 33 33 36 30 32 37 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 30 bytes of data Handling message for SAPI/TEI=0/127 TEI: 0 State 7 V(S) 8 V(A) 8 V(R) 8 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0 T200 0, N200 3, T203 1 [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21 a3 70 07 81 33 33 36 30 32 37 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 30 bytes of data Handling message for SAPI/TEI=0/127 - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI PRI error message
On 6/17/2010 6:47 AM, Gareth Blades wrote: Scott Stingel wrote: On 6/17/2010 2:12 AM, Gareth Blades wrote: Scott Stingel wrote: Hello- After configuring DAHDI and starting asterisk, I get the following message continuously on the Asterisk console: WARNING[2057]: chan_dahdi.c:4158 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! My card is a D410P configured for E1, only the first span is configured, and configuration snippets are as follows: From /etc/dahdi/system.conf: (auto configured, first span only shown:) # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 -- From /etc/asterisk/dahdi-channels.conf (included in chan_dahdi.conf): ; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 context = default group = 63 -- QUESTION: Shouldn't asterisk pick up from dahdi.conf that the signalling channel is 16? Why the error message? Thanks Scott The error means that when it tried using the D channel 16 it was not able to communicate. Probably a misconfiguration or you havent plugged the cable in yet. Thanks Gareth- I'm zeroing in on the pattern of this channel allocation problem: This is in a system which has both a D410P and B200P (OpenVox). When the D410P (wct4xxp driver) is configured by itself, there are no error messages and the PRI's work fine. However, when I also load the wcb4xxp driver to support the B200P in the same system, I start getting these error messages and the PRI spans no longer work. I'm going to further quantify these symptoms and will report back Maybe a channel ordering problem? If you have a 4 port analogue card for example then if that card is detected first you will find it is channels 1-4 and therefore your ISDN30 will be channels 5-19,21-35 and the D channel being 20. Yes, this was my thought too (wouldn't each BRI be 3 channels 2B+D though, so 6 channels for the B200P?), but despite much effort still unable to get it to work in configurations that have worked in the past with asterisk and bristuff.Have tried the PRI first and them the BRI first. Anyway, more experimentation is in order! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI PRI error message
Hello- After configuring DAHDI and starting asterisk, I get the following message continuously on the Asterisk console: WARNING[2057]: chan_dahdi.c:4158 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! My card is a D410P configured for E1, only the first span is configured, and configuration snippets are as follows: From /etc/dahdi/system.conf: (auto configured, first span only shown:) # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 -- From /etc/asterisk/dahdi-channels.conf (included in chan_dahdi.conf): ; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 context = default group = 63 -- QUESTION: Shouldn't asterisk pick up from dahdi.conf that the signalling channel is 16? Why the error message? Thanks Scott -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID's for Chatham, ON
I think there's an Asterisk-Biz mail list for this purpose.. -Scott On 5/29/2010 6:12 PM, Robert Augustyn wrote: Can anybody provide DIDs for Chatham, ON? Usage based preferred, but flat-rate is not an issue. Contact off list. Thanks for your time, Sincerely, Robert Augustyn -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenVox B200P and D410P under Asterisk 1.6
Hello all- My client has purchased these two OpenVox cards and I'm configuring a system with Asterisk 1.6. In the past I have used bristuff and libpri with older versions of Asterisk, but now I would like to upgrade to Asterisk 1.6. Question, should I be using mISDN or libpri for these cards when they are in the same system, or does DAHDI now support both cards under asterisk 1.6 reliably? I'm especially concerned about the OpenVox B200P as I haven't used it before. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVox B200P and D410P under Asterisk 1.6
On 5/27/2010 2:33 PM, Philipp von Klitzing wrote: Hi! Question, should I be using mISDN or libpri for these cards when they are in the same system, or does DAHDI now support both cards under asterisk 1.6 reliably? I cannot answer that question, but do stay away from mISDN if you can. Philipp OK thanks Philipp. OpenVox has been steering me toward mISDN for their B200P card, but I am reluctant given what I've learned so far. My experience is only with bristuff, but I had hoped to use the generic DAHDI. -Scott -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rockwell ACD - Take back and transfer
Hi all- I have a customer with a Rockwell Spectrum ACD. They wish to connect to an asterisk system using euro-ISDN circuits, and asked me if asterisk supports a feature that they call Take back and Transfer. This feature allows an IVR (asterisk in this case) to handle a call and then blind-transfer it back through the ACD to another extension- freeing up the asterisk channel. It's seems similar to the 2B Channel Transfer (2BCT) that is discussed from time-to-time on here, and in the Dev forum. I read that one of the libpri developers has recently added Q-Sig Path Replacement, which is described as a Q-Sig version of 2BCT. Anyone have an idea whether this will work with the Rockwell ACD? Thanks in advance, Scott Stingel www.evtmedia.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS Text Send working with BT Text in the UK??
Thanks Julian - will update and see if it works. regards, Scott Julian Lyndon-Smith wrote: We are using 1.4 trunk with sms - it got fixed recently (about 4-5 weeks ago I think). Julian Scott Stingel wrote: Hi all- In 2004, I set up a sms texting process for a UK customer, using the asterisk SMS command and BT's BT Text SMS facility. This has been running fine up until recently. A couple of weeks ago, I upgraded them from their old asterisk version (CVS 2004.07.25) to 1.2.9.1, and have been having trouble getting the SMS feature to work on this newer version. I'm connecting to BT via a BRI, running an updated bristuff. (was also running this configuration previously) I do note the differences called out in the documentation, mainly that smsq is used to set up parameters for the text to be sent, and I've changed my code appropriately. Here is what I try: smsq --motx-channel=Zap/g3/17094001 --motx-retries=0 0111222 Hello! This seems to start things happening, as I observe the following on the asterisk console: --- -- Attempting call on Zap/g3/17094001 for application SMS(0) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH Sep 27 20:47:51 NOTICE[13661]: channel.c:2455 __ast_request_and_dial: Don't know what to do with control frame 15 Channel Zap/7-1 was answered. Launching SMS(0) on Zap/7-1 -- SMS RX 93 00 6D -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- SMS RX 92 01 01 6C -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- SMS RX 92 01 01 6C -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- Channel 0/1, span 3 received AOC-E charging 0 units -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- Hungup 'Zap/7-1' Sep 27 20:47:59 NOTICE[13661]: pbx_spool.c:279 attempt_thread: Call completed to Zap/g3/17094001 --- From looking at the app_sms.c code, I seem to be connecting to BT ok, but it appears that the 92 code received from them indicates an error in the format. As other posts have suggested,I have tried the following: (a) going back to version 1.2.7.1 (same symptoms) and (b) increasing the wait for response delay (h-opause) -no effect either. I've also tried reverting to my 2 year old app_sms.c, which no longer compiles (as expected) Does anyone have asterisk SMS texting via BT working in the UK, using a recent asterisk version, and if so, can you please shed some light on this? Many thanks Scott Stingel www.evtmedia.com ___ --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 ___ --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 . ___ --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] E1 crossover system
Hi Paul- Just checking that you have a proper E1 crossover cable? (not a CAT5/6 crossover or patch cord - they are not the same) Crossover is wired as per: http://www.evtmedia.com/designersFAQ.htm#Make%20E1/T1%20crossover Most likely you would need a straight-thru cable for your application I would think, not a crossover, but I mention it anyway. Best regards, Scott Stingel evtmedia.com Paul Hales wrote: I am setting up an E1 crossover system for a customer, with a Siemens Hipath Officecom 150 system. And it's not working - I get a red alarm from the outside world, and a yellow from the PABX. Any ideas? We have tried E1 crossovers and straight cables on both connections, with no more luck. Frame type? Cabling? Timimg? Any ideas at all? Anyone? PaulH ___ --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 . ___ --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] SMS Text Send working with BT Text in the UK??
Hi all- In 2004, I set up a sms texting process for a UK customer, using the asterisk SMS command and BT's BT Text SMS facility. This has been running fine up until recently. A couple of weeks ago, I upgraded them from their old asterisk version (CVS 2004.07.25) to 1.2.9.1, and have been having trouble getting the SMS feature to work on this newer version. I'm connecting to BT via a BRI, running an updated bristuff. (was also running this configuration previously) I do note the differences called out in the documentation, mainly that smsq is used to set up parameters for the text to be sent, and I've changed my code appropriately. Here is what I try: smsq --motx-channel=Zap/g3/17094001 --motx-retries=0 0111222 Hello! This seems to start things happening, as I observe the following on the asterisk console: --- -- Attempting call on Zap/g3/17094001 for application SMS(0) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH Sep 27 20:47:51 NOTICE[13661]: channel.c:2455 __ast_request_and_dial: Don't know what to do with control frame 15 Channel Zap/7-1 was answered. Launching SMS(0) on Zap/7-1 -- SMS RX 93 00 6D -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- SMS RX 92 01 01 6C -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- SMS RX 92 01 01 6C -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- Channel 0/1, span 3 received AOC-E charging 0 units -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- Hungup 'Zap/7-1' Sep 27 20:47:59 NOTICE[13661]: pbx_spool.c:279 attempt_thread: Call completed to Zap/g3/17094001 --- From looking at the app_sms.c code, I seem to be connecting to BT ok, but it appears that the 92 code received from them indicates an error in the format. As other posts have suggested,I have tried the following: (a) going back to version 1.2.7.1 (same symptoms) and (b) increasing the wait for response delay (h-opause) -no effect either. I've also tried reverting to my 2 year old app_sms.c, which no longer compiles (as expected) Does anyone have asterisk SMS texting via BT working in the UK, using a recent asterisk version, and if so, can you please shed some light on this? Many thanks Scott Stingel www.evtmedia.com ___ --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] TBCT - Two B-Channel Transfer
Hi- A customer of mine running asterisk has inquired if asterisk currently supports TBCT, that is the ability of asterisk to transfer a call back up to the carrier's switch to complete a connection, freeing up the B-channels on the asterisk box. I saw a reference to this feature in the asterisk bounties section on the Wiki, and someone added a note saying that this might be possible in a 5ESS configuration. Has anyone actually used TBCT under asterisk? Thanks Scott Stingel Emerging Voice Technology, Inc. www.evtmedia.com ___ --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] Telephone line installation.
Manjit- I'm not sure what country you are in, but if you're in the USA, the telephone company will typically bring the connections to what they call the demarcation point. This is usually a punch-down block in the telephone closet for your building. They will usually offer to wire up telephone jacks wherever you want (called inside wiring), but usually at quite a high hourly rate. Regards Scott Stingel www.evtmedia.com Manjit Riat wrote: Thanks but I am pretty sure they won't do it. So there has to be a better way. -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 13, 2005 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Telephone line installation. They should have the answer: http://local.sprint.com/home/local/contact/contact_information.html On 4/13/05, Manjit Riat [EMAIL PROTECTED] wrote: We are going to be doing an asterisk install with 5-7 lines. So we are looking to get two TDM04B cards. Now I believe when you get your telco(Sprint, etc.) to install the lines they basically just leave the wires without jacks. Am I right? If so, then can we ask them to install the jacks or would we have to do them ourselves? ___ 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 ___ 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 . ___ 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] TE410P and X101P problem
LeeLee- Try configuring all 4 spans first and then the single channel (125) above that - works for me. Modprobe in the same order, then ztcfg. Regards Scott Stingel www.evtmedia.com Lee Lee wrote: Hi all I newly added a X101P into my asterisk that already have a TE410P running 2 E1s namely span1 and span2 I am unable to get * to recognized the new X101P after i did modprbe wct4xxp and then modprobe wcfxo. ztcfg -vv reported all 63 channels are configured but zttool tells me that span 1,2,3 are OK and X101P UNCONFIGURED. I do not have anything plug into span 3 below are what i have _zapata.conf_ [channels] context=default overlapdial=no signalling=pri_cpe switchtype=euroisdn pridialplan=unknown rxwink=125 echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.9 immediate=yes musiconhold=default group=1 channel = 1-15,17-31 busydetect=no group=2 channel = 32-46,48-62 busydetect=no group=5 signalling=fxs_ks channel=63 context=default _zaptel.conf_ span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 alaw=1-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 alaw=32-62 fxsks=63 loadzone=us _ztcfg -vv_ Channel 62: Individual Clear channel (A-law) (Slaves: 62) Channel 63: FXS Kewlstart (Default) (Slaves: 63) 63 channels configured. Do you Yahoo!? Make Yahoo! your home page http://us.rd.yahoo.com/my/navbar/sethp/*http://www.yahoo.com/r/hs ___ 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 ___ 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] Problem with X101P
Some questions: What country are you in? Is there anything else connected to the line from the PSTN? It sounds like you have a marginal condition, such as insufficient loop current perhaps. Do have any features, such as call waiting, on the line? Do you know how far you are from the central office? Do you have another line you can switch to and try the same card? Does the Red alarm occur at the moment the call is disconnected, or afterward? Regards Scott Stingel www.evtmedia.com Yusuf Iqbal wrote: Previously I have posted the same mail but no one answered me...Sorry for resending the mail. I have bought a Wildcard X101P for my Asterisk PBX. Now I can place and get calls through the lines/channel. Everything is okay but the problem is when I call outside through our PSTN line, after few minutes the connection breaks down. The same thing happens in case of incoming calls. I have checked my wiring and don't face that problem using direct connection. Whenever I call using that card, after few minutes I get a RED Alarm and if I reconnect the line, the Alarm is cleared. Therefore, I cannot continue my conversation through that line. Can anybody help me regarding this problem? // Express yourself instantly with MSN Messenger! MSN Messenger http://g.msn.com/8HMAEN/2728??PS=47575 Download today it's FREE! ___ 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 ___ 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] Sangoma VS. Digium
Peter Svensson wrote: Resellers are almost universally a useless money-sink. Most add no value at all, they are simply another logistics point. Distributors, on the other hand, are usually very knowlegable and are able to support their customers (the resellers) quite well. My advice: always *always* buy from as early in the channel as possible. Prices are better and the support is _way_ better. Of course, if you are not familiar with the problem space for which you are purchasing a solution then resellers can add a lot of value. Peter Hi Peter- In my experience, I've found that the biggest advantage to buying through a distributor is the availability of stock, in-house and available for rapid shipment. Distributors typically handle dozens of lines and do not have the manpower to be technically up-to-date on all products. I agree with you though about buying as high up in the channel as possible. For a small company like Digium, it makes perfect sense for them to provide the hardware support for their products directly, as they are the only ones who can do so practically until they are of a size to develop the huge amount of training materials, spare parts kits etc to allow distributors to provide any kind of support to the customers. Unfortunately, the volumes are probably not really there yet to justify this. Digium needs to take a hard look at it's support and hardware documentation, as well as size of its engineering staff. It seems to me that they are at a point of growth where a significant investment is required to allow growth to the next level. Regards Scott Stingel www.evtmedia.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] RE: Sangoma VS. Digium
Good question. I've had good luck with the Digium TE405P recently in a multiple T1 install. Don't let the discussion scare you too much! -Scott Stingel cmould wrote: Where is this discussion going. I am about to do an installation that will require t1 interfaces. I am new to the telephone world and found the original discussion useful. I need to know from a reliability and performance standpoint what is the better choice. Sangoma or Digium? ___ 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 ___ 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] My Sangoma Experience - Review
Bravo - nice writeup Matt! It concisely captures both the pros and cons. Seems that we really do have (or are close to having) a second source now - and all asterisk users will benefit in my opinion! Cheers Scott Stingel www.evtmedia.com mattf wrote: My Sangoma Experience in Asterisk: 2005-04-07 Having pushed my Digium Asterisk systems to their capacity many times and figuring out the limits of the Digium hardware I decided it was time to test an Asterisk-compatible Sangoma Quad T1/E1 card(AFT-A104u) to see if they live up to their hype of being more efficient than the Digium variety(T405P). I had talked with someone from Sangoma before at Astricon, but it was rather informal, he didn't have any literature and I was rather swamped at the time as it was. Then I saw a posting on the asterisk-users list about the claims that the Sangoma card does echo-cancelation better as well as using far less interrupts than Digium hardware(a big bottleneck with busy Digium systems). I emailed Sangoma(they are located in Canada) for a quote and quickly received a phone call from them. They were very interested in getting my feedback on using their quad port T1/E1 card with Asterisk and they quoted me a discounted price of $1190 US for the card(They said retail was $1700 US [Digium quad-cards are $1495 retail but you can get them through resellers for a couple hundred less]). The Sangoma card comes with a 30-day money back guarantee and a 3 year warranty. When I received the card I noticed a couple things right away, it was a very professionally packaged item and it came with 4 T1 cables in the box as well as documentation and all of the other pretty things you expect in a retail package. The second thing I noticed is that the card was compatible with a 2U form-factor(That's right, they crammed 4 T1/E1 ports together so it can fit in a 2U case vertically) This was achieved in-part because the ports are actually on a fixed daughter card, but it did bring up the thought that they could actually cram 6 ports on one of these cards :) Next I started to sort through the documentation and files on their FTP site. I noticed something I wish Digium cards had: User-upgradable firmware on the board(I have previously had to return an early version of the T410P Digium board to get a newer one with newer firmware on it). Let the installation begin. I started by downloading and installing Asterisk as usual(zaptel, libpri, asterisk[version 1.0.6]), then I downloaded and installed Wanpipe release 2.3.2 beta6. I could now see my card and went into the wancfg utility to configure my card. Here's when it stopped being a smooth experience. I tried installing it by the asterisk instructions found on the FTP site(which I found out later were out of date and incorrect) and eventually it all worked up until the final starting step. The drivers saw the card, but said nothing was connected to them which I thought was a strange problem since you don't have to have anything connected to a Digium card for Asterisk to fully startup. So I emailed tech support and walked through some reconfiguration steps and then after a few more emails back and forth it came out that they had a problem with D4/AMI signalling on a RBS T1(which they say they will have a fix for at some undefined time in the future). After switching the wanpipe config for the first span to B8ZS/ESF with a PRI T1 I was able to run ztcfg and asterisk. I placed some test calls and all went well, at least until I tried hooking up a live RBS(Robbed-bit, 24 full channels not PRI) EM Wink T1. It turns out that the guys at Sangoma have never had a customer that used EM Wink start and accordingly they have never tested their cards with it, and of course it didn't work. So another email and call to Sangoma and they started working on a fix. Two days later they added a Wink for wink start T1s and sent me a new version of the software. I loaded it and it worked, but all audio and call detects stopped working if I tried to use more than 10 of the RBS T1 channels, so back to Sangoma for another new driver version. After a few days, and a few more driver versions, they came up with one that seemed to fix all of the problems I was having before so I did my simple stress test of picking up, hanging up and redirecting to meetme of about 52 Zap lines and all went well. Now on to the performance testing. For a performance test, I swapped out an identically configured machine that had a Digium T405P with my test machine and put it live in company inbound/outbound call center during off-hours to test(This server usually handles over 20,000 calls in/out a day with lots of recording going on across T1s, SIP phones and some IAX2 trunks). This server has two RBS T1s, one PRI T1 and one Channel Bank. I placed a test call out of the channel bank through the PRI and then started automated calls from the two RBS T1s to go into meetme conferences. The performance test ran great and it did prove
Re: [Asterisk-Users] [OT]: Wiki Etiquette
Good question. Maybe it would be better to politely post an opposing view right below the seemingly wrong information. My only experience in editing other people's postings was when I corrected a poster that had incorrectly put his company at the top of the list of asterisk consultants, out of alphabetical order as the list requests. My edit was promptly re-edited, so I gave up! Anyway, just one of the risks of an open Wiki! regards Scott Stingel Emerging Voice Technology www.evtmedia.com Sean Kennedy wrote: Hi folks, I recently registered with the wiki site to fix a few things I've noticed, and I had a question: Is it proper to delete other people's additions if they are obviously incorrect? My main concern is for the content, which is ( well, was ) false. On the other hand, I do not want to start a pissing match with anybody because of bruised egos. Further, in some cases that I've seen, the OP might have a valid point, but it is not one shared by the general populous. In my mind, that view should be respected, but on the other hand, I feel there should be a correction to the wiki regarding it. Any input on this would be greatly apprecaited. ___ 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 . ___ 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] zaptel.conf digium and quadBri together (e1 and isdn together)
Hello Victor: Are you using the QuadBRI from Junghanns? If so, this is a configuration that works for me in the UK, one single E1 board and the QuadBRI. As I recall, the order of loading is important (see way below): ZAPTEL.CONF loadzone=nl defaultzone=nl # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami # E1 definition: span=5,0,0,ccs,hdb3,crc4 #BRI's: bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 #E1: bchan=13-27,29-43 dchan=28 - ZAPATA.CONF [channels] ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp ; define 4 BRI's: pridialplan = unknown prilocaldialplan = unknown echocancel = yes context=incoming group = 1 channel = 1-2 group = 2 channel = 4-5 group = 3 channel = 7-8 group = 4 channel = 10-11 group = 9 ;and the E1 pridialplan = unknown signalling=pri_net - this is because I'm connecting to a Dialogic board in another system, not the PSTN.. Would normally be pri_cpe channel = 13-27,29-43 - IN YOUR SYSTEM REBOOT ROUTINE: #following for Quad BRI system: cd /usr/src/bri/bri-stuff.0.1.0-RC2g/qozap modprobe -v zaptel /var/log/asterisk/modprobe.log sleep 1 insmod -v qozap.o /var/log/asterisk/modprobe.log sleep 1 # following for single E1 system modprobe -v wct1xxp /var/log/asterisk/modprobe.log sleep 4 #ztcfg -vv /var/log/asterisk/modprobe.log - NOTE THAT ztcfg commented out, Junghanns issue #sleep 3 startast sleep 3 GOOD LUCK Scott Stingel www.evtmedia.com - Victor Alvarez wrote: Hello, I have a machine with two cards installed, one digium that gives e1 connectivity and one quadBri for the ISDN line. I can use them independently. I have one zaptel.conf and one zapata.conf for each card. I would like to work with them at the same time and I am not sure about how could I do it (if it is possible!). My first attempt wasn't very successful. I made the config files getting the info from the one for isdn and the one for e1. Ztcfg configures the 136 channels without problems but I can't use the e1 channel. A curious thing is that, with the same configuration, I can use the e1 channel. Any help/suggestion is more than welcome. Thanks. ___ 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 ___ 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] TE405P and Dell Poweredge 6450 Incompatible?
Hi Matthew- I think there are TE410P compatibility issues with other motherboards as well. Google the archives (site:digium.com) under HP Proliant G4 for example, as I remember some problems there. This response from Digium tech support, if quoted accurately, is not acceptable to customers for obvious reasons. If there is a known compatibility issue with some chipsets, I think it would be good idea for Digium to publish them. Otherwise IMHO, they should offer to accept your return of the TE410P so you can pursue an alternative. Perhaps a call to the marketing people at Digium would be in order? It's just bad PR for the company when these things happen. Please keep us up to date on what happens. regards Scott Stingel www.evtmedia.com [EMAIL PROTECTED] wrote: Hey gang, Just pulled out our brand new $1,500 TE405P and put it into a Dell Poweredge 6450. Nothing. Card not recognized nor listed under lspci. Tried the card in every slot it would fit in. Even took out existing, working cards and put the 405 in those slots and still nothing. Tried the card in a Poweredge 2450. Bingo, although it is listed as Unknown device. Called Digium. Their answer Since the card works in the 2450, you will have to use it in the 2450. The card is good. My response was the 2450 is a single P3 500 and no where near powerful enough to handle 92 simultaneous PRI-g729 calls. Has anyone else out there had success/failures with a TE4XX card in Dell Poweredges? I've been on hold at digium for a while now and I get the feeling that Digium is gonna say too bad sucka..the card works..*click* but if enough people have a problem with this card in Dell Poweredges, then perhaps they will take a closer look and figure out the problem. -Matthew This message was sent using IMP, the Internet Messaging Program. ___ 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 . ___ 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
[Asterisk-Users] RE: Digium ISDN card
(thread moved from asterisk-dev) Hello Wai- You can find out a lot about the digium cards and asterisk software on the Wiki, which is at: http://www.voip-info.org/tiki-index.php?page=Asterisk Briefly, though, asterisk implements things like DTMF and voice detection using the host processor's hardware/software. This makes the cost of the interface card much lower than, say, a Dialogic, but limits to some extent the number of channels you can handle with one processor. With the size of the processor you mentioned, for most applications you could probably handle between 3 and 4 full E1 spans, but heavy transcoding might cut that number down somewhat. Test before you commit! Hope this gets you started. Regards Scott Stingel www.evtmedia.com Wai Wu wrote: Hi all, My first post here. We will be implementing an * solution with Digium Pri cards. I have a few questions. First, what is the largest * system anyone has put together with a P4 3.4GHz with a 1GByte of ram? My second question is regarding to the Digium Pri card. On the Digium site, I notics those cards are half length cards and they are quite small. As dtmf detection, and active voice detection is concerned, are they done on the card itself or they are done in *? Regards ___ 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] Sangoma VS. Digium
I think the telecom market is so huge that it can easily support several hardware suppliers - and all of them can be successful if they make a good product. It can be good for Digium (and ultimately for us) that Sangoma is providing some competition, as it will drive Digium to new levels of performance and reliability. -Scott Stingel www.evtmedia.com Andrew Kohlsmith wrote: I respectfully disagree. Sangoma's voice capabilities are no less and no more mature than Digium's voice capabilities. I use cards from both Sangoma and Digium. Both seem to work well but (and it does pain me to say it, it really does) Digium's cards seem FAR more finicky about the type of hardware they'll run reliably on. Sangoma's cards you can pretty much throw into any system and they work. Shared interrupts and oddball PCI chipsets included. I do believe, however, that this is merely a driver issue. If I were a more competent driver programmer I would certainly dive into this headfirst. -A. ___ 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 . ___ 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
[Asterisk-Users] RE: TE410P Loadtest problem
(Thread moved from -dev)
Hi Ma-
You probably can do this with a 3GHz+ processor, but I don't have experience
with such a busy voicemail server - perhaps someone on here will...
Regards,
Scott Stingel
www.evtmedia.com
Ma Zhiyong wrote:
Thanks Scott.
I sure read your note on wiki before I do the loadtest. Very helpful.
My application is a voicemail system. So I expect all channels can work
well even under the scenario that all of them is used at the same time.
So Can I do this with using a more powerful processor?
What kinds of suggestion do you have?
Thanks.
--
Scott Stingel wrote:
Hi-
Last year, I also did extensive tests on the TE410P, running loopbacks
within one box (a P4 at 2.8GHz) and also between two similar boxes, each
with a TE410P running all spans.
Although my tests didn't involve the voicemail application like yours, I
found that one box can barely keep up with all four E1's running full
call loads (with short duration calls), and that two TE410P's in one box
definitely way overloaded a box. I demonstrated this scenario to Mark
Spencer at Digium at the time (about a year ago now).
My theory was that new call setup contributed to call load, not just the
number of simultaneous connections. You *might* be OK running 4 spans in
a box if the calls were normal conversations and not doing much else.
Perhaps the voicemail app (recording in particular) presents a heavy load.
I also put some notes about my tests in the asterisk Wiki at the time,
under the Dimensioning section.
Regards
Scott Stingel
www.evtmedia.com
Ma Zhiyong wrote:
Hi,ALL
I made a voicemail box based on * and one TE410P card on Redhat
platform. My server worked with SuperMacro P4SCI motherboard, P4 2.4c
CPU and 1G memory. Things couldn't be better. So I did the loadtest.
I use two E1 ports as pri_cpe and other two ports as pri_net. Then I
connected them b2b and use autodialed outgoing calls to play sound in
one channel and record the sound in another correspondingly.
When I made 50 calls, that meant 100 channels was used. I could found
msg*.wav files in INBOX directory of 50 vm users. And the record files
was good. I check the CPU time use top command just like the list below.
#top
PID USER %CPU %MEM TIME CPU COMMAND
3715 root 9X.X 1.5 1:20 0 asterisk
1 root 0.0 0.0 0:05 0 init
2 root 0.0 0.0 0:00 0 keventd
3 root 0.0 0.0 0:00 0 kapmd
But when I made 60 calls, that used all 120 channels. It didn't work
well. I couldn't found all 60 msg*.wav in INBOX directory of 60 vm
users. And some of the existed msg*.wav was not completed.
I think that means some of the calls lost.
Do that mean my CPU is not good? Or some other reasons?
Anybody has similar experience?
Thanks.
BTW below is my dialplan and callfile.
[from-te410p]
exten = 5,1,Answer
exten = 5,2,Wait,1
exten = 5,3,Voicemail(u${CALLERIDNUM}) ; just test
exten = 5,4,Hangup
[loadtest]
exten = 0,1,Answer
exten = 0,2,Wait,3
exten = 0,3,Playback(DTMF-pound)
;DTMF-pound is a DTMF pound tone
exten = 0,4,Playback(demo-instruct) ; Play some instructions
exten = 0,5,Playback(DTMF-pound)
exten = 0,6,Hangup
#generate call files use a shell.
CALL_TMP_DIR=/var/spool/asterisk/tmp
CALL_SPOOL_DIR=/var/spool/asterisk/outgoing
make_callfile()
{
# the load test call files using voicemail box 5501-5530 5601-5630
if [ x$1 != x ];then mailbox=$1; else return 1; fi
if [ x$2 != x ];then channel=$2; else return 1; fi
# note use of '-' in '-EOF1' - Escapes tab at beginning of lines
CALLFILE=$(cat -EOF1
Channel: ZAP/$channel/5
Callerid: $mailbox
MaxRetries: 2
RetryTime: 120
WaitTime: 30
Context: loadtest
Extension: 0
Priority: 1
EOF1)
echo $CALLFILE $CALL_TMP_DIR/callfile$mailbox-$(date +%s)
}
i=5501
j=5530
while [ $i -le $j ]
do
make_callfile $i g4
`mv callfile$i-* $CALL_SPOOL_DIR`
echo file$i generated!
i=`expr $i + 1`
done
i=5601
j=5630
while [ $i -le $j ]
do
make_callfile $i g2
`mv callfile$i-* $CALL_SPOOL_DIR`
echo file$i generated!
i=`expr $i + 1`
done
--
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Re: [Asterisk-Users] TE110P module woes
Just to confirm that you also powered down and up? I've no experience with the TE110, but this is a known problem with the TE405 and TE410. They apparently can get locked up, and only a power cycle will clear it. Regards Scott Stingel www.evtmedia.com Alfredo Sola wrote: Hi, I have been using asterisk for a couple of months now and for thee most part, I love it. However, I'm having a problem with the drivers of the Digium TE110P. I have tried both the Debian package and the CVS. I have tried several kernels, and am now at 2.6.11. This has been working before (with 2.6.8.1), but after a reboot it stopped working and I am not able to consistently make it work or fail. I have make clean, make and make install, no complains from make. The zaptel module loads fine and says so: Zapata Telephony Interface Registered on major 196 But the module for the TE110P fails. If I only modprobe it, it loads silently; but the moment I execute ztcfg, I get: ZT_SPANCONFIG failed on span 1: No such device or address (6) If I ask for more verbose errors, I get: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) Any ideas? ___ 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] Sangoma Cards
From looking at the description, it seems that the Sangoma card (at least the quad version) *may* have a more robust hardware buffering mechanism than the TE4xxP series. If so, this might help solve some of the load-related issues that my customers have experienced in very large systems. Hope Digium takes note and makes their own improvements! I'm a loyal Digium customer and reseller and would like to stay that way... Cheers Scott Michael Bielicki wrote: There are no DSP's on the sangoma cards, who gave you that idea ? The nice thing about those cards are: selective echo cancellation per span around 25% less interrupts created so less load auto select on 3.3V/5V and some other engineering details plus the warranty of 3 years. But no DSP's. cheers Michael On Sat, 26 Feb 2005 23:28:38 +0200, Calin Serbanescu [EMAIL PROTECTED] wrote: Hello list, I need a few words about the difference between sangoma quad E1 cards w/dsp vs. digium tormenta2 compatible cards. Does * really make use of the dsp's on these boards(sangoma)? How many % CPU do they each need (sangoma vs. digium)? Unfortunatelly i do not have the sangoma cards yet, they're on their way with DHL, but i'm very curious about them. Thanks for your time, Calin. ___ 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 ___ 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] Difference between E1 and PRI
Alistair- Good writeup! Question regarding Q.SIG: Can it be used to solve the problem of signaling a remote switch to take a call back and extend it to another channel instead? This, as you know, is always a challenge when using IVR in a call centre environment, when one wants to extend an IVR call to a live operator without holding up channels in the IVR. Regards, Scott Stingel Emerging Voice Technology, Inc. www.evtmedia.com Alistair Cunningham wrote: Eric, E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe, which can be used for data, or can be split into 32 64Kbps telephone channels, or a mixture. If used for telephone channels, 30 of these channels can carry one telephone conversation each, and 2 carry signalling and timing information. T1 is similar to E1. It is used in North America. It is 1.544Mbps, and can carry 24 telephone channels, each of which can carry a telephone conversation (but see below). There are a number of protocols which can run on top of E1. Some of these are called CAS, Channel Associated Signalling. Examples are FXS loop start and EM wink start. They provide information such as the number that was called, and what state the call is in. They're limited in what information they can carry, and are slow to set up. A more modern protocol which overcomes these problems is ISDN. On E1, EuroISDN is the standard. On T1, there are different standards from different providers. DMS100, DMS250, NI1, and NI2 are common examples. ISDN uses one channel (called the D channel) for signalling call information. On E1, this is one of the 2 signalling channels, leaving 30 channels for voice (called B channels). On T1, there aren't any spare signalling channels, so one of the voice channels is used, leaving 23 B channels for voice. A PRI (Primary Rate ISDN) is simply an E1 or T1 with ISDN on top of it. ISDN gives fast, reliable call setup and hangup detection, and detailed information about the call. In the UK, PRI is also called ISDN30. An important extension to ISDN is Q.SIG, which provides extra signalling information that is used when connecting PBX systems. An alternative to PRI is BRI (Basic Rate ISDN), which is a cheaper system for small offices. It has 2 64Kbps B channels for voice, and 1 16Kbps D channel for signalling. It is sold as an alternative to analogue telephone lines. IN the UK, it is also called ISDN2e. I hope this answers your question! My company offers commercial support and installation services for PRI and Asterisk if you need help for specific scenarios. This email may form the basis of a future Integrics Tip. See: http://integrics.com/tips/ Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ 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 ___ 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 . ___ 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] Problem connecting a TE410P to an E1/IP equipment
To test the TE410P itself, you can construct a simple cross-over cable by hacking up a short CAT-5 cable as you describe: 1 - 4 2 - 5 4 - 1 5 - 2 Note that a CAT5 crossover cable will not work. Once you've done this, set up two spans on your TE410P as you've done it, except that one of them should be CPE and the other one NET in zapata.conf. Now calls originated on any of the 30 channels on one span will appear as incoming on the other one. I don't think the clock source as set in zaptel makes any difference for these purposes, as it's specifying the clock source for functions like meet-me, not the framing clock which is derived from the E1 signal itself. One last thing is to check the jumpers on the card itself to make sure they are set for E1. (doesn't look like this is the problem tho) And, finally, actually shut the system all the way down, including power, and reboot to make sure new settings are loaded into the TE410. I think this is a design shortcoming in the card that it's possible for it not to clear it's settings even on a restart (without power down) With the setup above, you should get green lights on the connected spans, and should be able to send calls from one to the other. By the way, to initiate outbound calls use the .call file facility (see the Wiki) Good luck! Scott Stingel President Emerging Voice Technology, Inc. Palo Alto California and London England www.evtmedia.com Johan Bilien wrote: I guess I need some special equipment to do the tests, right? Unfortunately I don't have any such tool :( Maybe I can try to connect the card to some other E1 equipment. One question: I'm not sure if this equipment requires a cross or a straight cable. I tried with a straight one, and since I got the green LED I assumed it was correct. But could it be that I need to cross the cable? I mean send pin 1 and 2 to 4 and 5? If so would the green LED be lit? Thanks, Johan. Thanks, Johan. ___ 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 . ___ 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] Canadian DIDs...
you may not be aware of the asterisk-biz mailing list, which is probably more appropriate for a discussion like this. you'll find many VoIP termination vendors hang out there too. Regards, Scott Stingel Mohit Muthanna wrote: Have you used them before? Do they provide commercial grade service? On Tue, 22 Feb 2005 10:08:57 -0500, Nabeel Jafferali [EMAIL PROTECTED] wrote: Anybody know a good IAX provider for Canadian DIDs? I currently use Xetricom for Toronto DIDs (C$7.50 each). I also know of someone who can provide a Toronto DID with unlimited* GTA calling for C$20. Nabeel ___ 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] te405P and german PMX
Just checking, but if you're in Germany, don't you want E1 rather than
T1 settings? Not sure what a PMX is.
If you do indeed want E1, then something like the following would be
used in zaptel.conf.. Note also that you have to change the hardware
jumpers on the board (4) to the E1 position.
#note, may need to add ,crc4 to end of span lines:
#first quad board
span=1,1,0,ccs,hdb3
span=2,0,0,ccs,hdb3
span=3,0,0,ccs,hdb3
span=4,0,0,ccs,hdb3
#first quad board
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109
Hope this helps!
regards,
Scott Stingel
President
Emerging Voice Technology, Inc.
www.evtmedia.com
Sören Malchow wrote:
Hi all,
i am stuck with the configuration of asterisk
- modules are loaded ( zaptel and wct4xxp )
- i have zaptel.conf configure, output of ztcfg -vv
--- snip --
/rapid:~# ztcfg -vv/
/Zaptel Configuration/
/==/
/SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)/
/SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)/
/SPAN 3: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)/
/SPAN 4: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)/
/Channel map:/
/Channel 01: E M (Default) (Slaves: 01)/
/Channel 02: E M (Default) (Slaves: 02)/
/Channel 03: E M (Default) (Slaves: 03)/
/Channel 04: E M (Default) (Slaves: 04)/
/Channel 05: E M (Default) (Slaves: 05)/
/Channel 06: E M (Default) (Slaves: 06)/
/Channel 07: E M (Default) (Slaves: 07)/
/Channel 08: E M (Default) (Slaves: 08)/
/Channel 09: E M (Default) (Slaves: 09)/
/Channel 10: E M (Default) (Slaves: 10)/
/Channel 11: E M (Default) (Slaves: 11)/
/Channel 12: E M (Default) (Slaves: 12)/
/Channel 13: E M (Default) (Slaves: 13)/
/Channel 14: E M (Default) (Slaves: 14)/
/Channel 15: E M (Default) (Slaves: 15)/
/Channel 16: E M (Default) (Slaves: 16)/
/Channel 17: E M (Default) (Slaves: 17)/
/Channel 18: E M (Default) (Slaves: 18)/
/Channel 19: E M (Default) (Slaves: 19)/
/Channel 20: E M (Default) (Slaves: 20)/
/Channel 21: E M (Default) (Slaves: 21)/
/Channel 22: E M (Default) (Slaves: 22)/
/Channel 23: E M (Default) (Slaves: 23)/
/Channel 24: E M (Default) (Slaves: 24)/
/24 channels configured./
/rapid:~#/
--- snip --
and on asterisk CLI zap show channels giv e the following
--- snip
/*CLI zap show channels/
/ Chan Extension Context Language MusicOnHold/
/ pseudodefault/
/ 1default/
/ 2default/
/ 3default/
/ 4default/
/ 5default/
/ 6default/
/ 7default/
/ 8default/
/ 9default/
/ 10default/
/ 11default/
/ 12default/
/ 13default/
/ 14default/
/ 15default/
/ 16default/
/ 17default/
/ 18default/
/ 19default/
/ 20default/
/ 21default/
/ 22default/
/ 23default/
/ 24default/
/*CLI/
snip
- i also have an outgoing extension defined
exten = _9.,1,Dial(${TRUNK}/${EXTEN:1})
but when i try to dial, there is the following message on the console
snip
/-- Executing Dial(SIP/90-1765, Zap/1/040432969547) in new stack/
/Urgent handler/
/Jan 22 20:41:16 NOTICE[1114446768]: app_dial.c:743 dial_exec: Unable
to create channel of type 'Zap'/
/ == Everyone is busy/congested at this time/
/-- Executing BackGround(SIP/90-1765, tt-allbusy) in new stack/
/Urgent handler/
/-- Playing 'tt-allbusy' (language 'de')/
/Urgent handler/
snip
any suggestions what is wrong ?
thanks
soeren
Soeren Malchow
Head of Central Technical Services
Interone Worldwide GmbH
Schulterblatt 58
20357 Hamburg
T +49.40.43 29 69 - 547
F +49.40.43 29 69 - 90
mailto:[EMAIL PROTECTED]
http://www.interone.de
NOTE: Information contained in this message is confidential and may be
legally privileged. If you are not the adressee indicated in this
message (or responsible for the delivery of the message to such
person), you may not copy, disclose or deliver this message or any
part of it to anyone, in any form. In such case, you should delete
this message and kindly notify the sender by reply Email. Opinions,
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Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario?
Andrew Kohlsmith wrote: Thunderbird, Eudora, hell even Pine I think. Thunderbird works very well but you have to enable it, since it doesn't do it by default. View - Sort by - Threaded -A. ___ 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 . I've switched fairly recently. Thunderbird is now pretty stable, and does threads well. It also has very good spam filtering built in. If you're used to Outlook, thought, you may miss the nice integration with your calendar, but you certainly won't miss paying for it! -Scott Stingel www.evtmedia.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] Prefered server hardware
-48m volt power is often used in telco central office environments, where the C.O. provides a huge amount of battery-backed up power to the switches and to power the local loops in the event of an AC power failure. Regards Scott Stingel Emerging Voice Technology, Inc. www.evtmedia.com Erick Perez wrote: This question is for my own knowledgei have no experience on this electrical area. why do you want to run -48vdc equipment? what's the advantage of doing that? On Tue, 18 Jan 2005 13:58:59 +0100, Daniel Nyström [EMAIL PROTECTED] wrote: What server hardware would you recommend for an Asterisk system which are really critical? The additional hardware will probably be two digium TE110P cards, and an Adit 600 platform. If it's possible to run on -48VDC, It would be great! Are there any experiences with any HP or FujitsuSiemens systems? Or other complete server systems? Thanks! BR Daniel Nyström ___ 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 ___ 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] sip to h.323
You can read all about it, and find out where to download at: http://www.voip-info.org/tiki-index.php?page=Asterisk Yes, it supports both SIP and H.323 Cheers Scott Stingel sai latha wrote: Hello, Happy New Year where u r downloaded the asterisk server please tell me.Iam searching the asterisk server site in google but i dint get this server u please tell me the site for me Is only for sip to sip or sip to h.323 please tell me Thank u Bye Sailatha ___ 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 . ___ 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] PRI Errors (HDLC Abort (6) on Primary D-channel)
Andrew- Thanks for posting your update and troubleshooting checklist.Most people on the forum don't take the time to re-post when a problem has been resolved - but that's the thing that helps people the most!. regards Scott Stingel President Emerging Voice Technology, Inc. www.evtmedia.com Andrew McRory wrote: UPDATE. The circuit has run clean since the 7th. It seems the telco found a problem after all but they can't tell me what they did... cause they think it was fixed as part of another case. Grrr. Well, I never doubted the T400P too much but I went ahead and bought a different card just in case. Haven't had to use it so far. One of the biggest reasons I did not think the Wildcard was at fault is that the only the LEC PRI span would crash. The other spans stayed up throughout the ordeal. Also the errors followed the PRI no matter what port it was plugged into. For anyone else seeing these errors, here is a checklist of things to look for: 1) Run the latest STABLE or CVS release of asterisk/zaptel/libpri If your problems started after an upgrade, revert to the revision that used to work. 2) Check and recheck your dialplan for errors. Make backups often so you can revert to a known good version. 3) Verify the line framing in /etc/zaptel.conf 4) Verify the switch type in /etc/asterisk/zapata.conf 5) Wildcard must be on its own IRQ. Example: [EMAIL PROTECTED] asterisk]# cat /proc/interrupts CPU0 CPU1 0: 52175299 0 local-APIC-edge timer 1: 7513 7309IO-APIC-edge keyboard 8: 0 1IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 10: 0 0 IO-APIC-level usb-ohci 15: 19 3IO-APIC-edge ide1 18: 260753631 260932719 IO-APIC-level tor2 20: 815258 827358 IO-APIC-level ide2, ide3 24: 4 8 IO-APIC-level mvSata 27: 836366 849040 IO-APIC-level eth0 28: 38 39 IO-APIC-level aic7xxx 31:24291412497473 IO-APIC-level eth2 NMI: 0 0 LOC: 52176184 52176197 ERR: 0 MIS: 12907 NOTE: I ran for over 6 months on a system that shared the IRQ with the Ethernet and USB - just to see what would happen. It worked very well but when the errors started, this was the first thing I HAD to fix. Sharing an IRQ with Wildcard is NOT recommended. Put the card on it's own IRQ! With todays highly integrated motherboard's, it's getting to where you can't buy one without an APIC - unless it is a cheapie. NOTE 2: To enable APIC you have to compile it into the kernel or if running FC1 kernel you must use the SMP version. I am not sure why APIC is not in the standard FC1 kernel. Planning to research this further but have not had time to so far. 6) Motherboard should support APIC (advanced programmable interrupt controller) Use quality motherboard (no el-cheapo chipsets) Use quality memory (run memtest86 to verify) Use quality peripherals 7) Compile a Kernel from plain sources ftp://ftp.kernel.org/pub/linux/kernel NOTE: The _Fedora_Core_1_ nptl.2199 kernel available on my site has been used here with great success. Can't speak for other precompiled sources since I haven't used them. Also your hardware may not work as well as mine with this kernel. 8) Make sure you have good power / power supply. Run a UPS! 9) If a multiport card, move the PRI/T1 span to another port to see if the errors follow. 10) If you have extra equipment to install on an unused span, configure it and see if it crashes at the same time the problem circuit does. If all circuits crash at the same time that would indicate a hardware problem such as IRQ or noise on the PCI bus (bad motherboard or PCI controller??). If all the above checks out these Red Alarms and HDLC errors are most likely due to problems in provisioning the circuit. At this point, become the Telco's biggest squeaky wheel or wait weeks for them to fix it by accident, if at all. Once more here is my configuration and it works :-) (T400P) -- ASTERISK -- SIP/VoIP/etc. | LEC-PRI - (Port1) (Port2) --- Max 4 / V.90 Dial-up users (Port3) --- Microcom isPorte / Fax Test (Port4) --- unused / test Thanks to all who offerd help!! I hope to not have to post about this error again! ___ Asterisk-Users mailing list Asterisk-Users
Re: [Asterisk-Users] TE410P problem (Looping UP Span 1...)
Sid- Try connecting one port to another. Note that one of the ports must be set up as cpe and the other as net in zapata.conf when you loop them together like this. A suitable crossover cable for testing can be constructed by cutting up a CAT 5 cable, and connecting: Pin 1 -- Pin 4 on the other end Pin 2 -- Pin 5 Pin 4 -- Pin 1 Pin 5 -- Pin 2 You should get green's on both the connected channels if your zaptel and zapata configurations are ok, and if you've run both modprobe and ztcfg as documented. Good luck Scott Stingel President EVT, Inc. www.evtmedia.com Sid wrote: Hi list, We have been trying to configure a Quad Span T1 card in a system running RH9. We have followed the instructions in the Wiki and searched the mailing lists, but so far havent got any success. Cable is connected to the first span, and module is loaded. Without loading the module the LED glows in red colour, but the moment we load module, it goes off. (No red or green) . We ran zttool and tried to run a loop test, but zttool simply hung with the message 'Looping UP Span 1...'. We had to terminate zttool with 'kill'. Here is the output of the lsmod command. Can someone shed some light on this? Thanks, -Sid Module Size Used byNot tainted wcusb 20128 0 (unused) wct4xxp54272 0 (unused) zaptel182432 0 [wcusb wct4xxp] tail -f /var/log/messages Jan 6 14:54:32 localhost kernel: TE410P: Launching card: 0 Jan 6 14:54:32 localhost kernel: TE410P: Setting up global serial parameters Jan 6 14:54:32 localhost kernel: Found a Wildcard: Wildcard TE410P-Xilinx Jan 6 14:54:32 localhost kernel: usb.c: registered new driver wcusb Jan 6 14:54:32 localhost kernel: Wildcard USB FXS Interface driver registered Jan 6 14:54:33 localhost kernel: Registered tone zone 0 (United States / North America) Jan 6 14:54:33 localhost kernel: TE410P: Span 1 configured for ESF/B8ZS Jan 6 14:54:33 localhost zaptel: Running ztcfg: succeeded Jan 6 14:55:07 localhost kernel: TE410P: Span 1 configured for ESF/B8ZS Jan 6 14:55:07 localhost kernel: Registered tone zone 0 (United States / North America) Do you Yahoo!? Yahoo! Mail http://us.rd.yahoo.com/mail_us/taglines/virus/*http://promotions.yahoo.com/new_mail/static/protection.html - Helps protect you from nasty viruses. ___ 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 ___ 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
[Asterisk-Users] Asterisk consultant wanted - S. California
Hello- I have a client in Orange County California who will soon need some consulting assistance with their new asterisk system. I've been asked to help them find someone. Skills needed would be, in order of importance: Basic experience configuring and using asterisk, coding experience in Perl, experience with MySQL or equiv., and a knowledge of telephony terminology and technologies. Would be very nice for the consultant to be located in Southern California to meet with customer occasionally. I have developed and delivered a working prototype of the system to their spec., but an increasing workload prevents me from carrying it much further. A number of customized (non-PBX) features will make this an interesting system to work on. 'C' coding or changing the asterisk internals should not be necessary as far as I can tell. Please contact me OFF-LINE (ie: NOT on this mailing list):scott at evtmedia.com, ie: do not reply to this, just send me a new email and please put: asterisk consulting or something in the subject line so I can see it among the spam! Thanks Scott Stingel President Emerging Voice Technology, Inc. www.evtmedia.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] HDLC Bad FCS (8) HDLC Abort on TE410P
Hi Roberto- About a year ago, I ran extensive loopback tests of the kind you described. I used various processors and motherboards, and used Fedora Core 1 with 2.4 kernels. (See the asterisk-users message archives and the wiki for more info). I got similar results to yours, except that I had no IRQ misses. I think that the underlying problem is that the processor is not quite keeping up with the interrupt service. My customer likes HP's, so I ran it on a couple models and found that I had to move up to the Proliant DL360 (3.4 Ghz Dual Xeon) before I could get absolutely trouble-free operation with all 120 channels running. At the time, the theory seemed to be that a lot of call setup's, not just overall call load, contributed to the problem, and that asterisk's recovery from frame errors was less than ideal and introduced a bunch of overhead. Anyway, I gave up looking for perfection and just use a very fast processor when I want to handle 120 channels with no errors. Best regards Scott Stingel President Emerging Voice Technology, Inc. www.evtmedia.com --- [EMAIL PROTECTED] wrote: Dear All, we have installed a TE410P card on a Dell Poweredge 1750 running a slackware 10 with 2.4.26 Kernel. Then we have made two loops on the card and we have configurated all the 120 channels. Our goals was to perform some stess tests even if in this scenario we used the same box as generator and target. The stress test comprised to generate up to 60 calls at the same time by placing a file in the /var/spool/asterisk/outgoing directory. The call included even the Play of a demo message. We did not face any problem up to 30 calls at the same time. Then the following notices came up: Jan 4 16:53:37 NOTICE[8621]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jan 4 16:53:37 NOTICE[8621]: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Increasing the traffic up to 50-60 calls at the same time it caused more serious problems such us the respawn of the channels. We have also noticed: cat /proc/zaptel/1 Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 HDB3/CCS/CRC4 ClockSource IRQ misses: 5 We have tried then to upgrade the box to the kernel 2.6. We have obtained less mistakes on the HDLC issues and better performance but we had to give up due to the fact that udevd caused to hug the machine when typing the modprobe of the card. Regarding HDLC Abort issues, Have you experienced such a problem? Do you think is a SW bug? Do you think that is it possible to reduce it by upgrading the kernel with preemtive patch? Our zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15, 17-31, 32-46, 48-62, 63-77, 79-93, 94-108, 110-124 dchan=16, 47, 78, 109 loadzone=it defaultzone=it zapata.conf [channels] ; ; GROUP 1 switchtype=EuroISDN overlapdial=yes usecallerid=yes pridialplan=UnKnown hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no ; group = 1 context=in switchtype=EuroISDN signalling=pri_cpe pridialplan=UnKnown echocancel=yes channel = 1-15, 17-31, 63-77, 79-93 ; ; GROUP 2 - ; switchtype=EuroISDN usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no ; group = 2 context=out signalling=pri_net channel = 32-46, 48-62, 94-108, 110-124 more /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 15106973 0 0 0IO-APIC-edge timer 1: 1534 0 0 0IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 8: 1 0 0 0IO-APIC-edge rtc 12: 0 0 0 0IO-APIC-edge PS/2 Mouse 15: 5 0 0 0IO-APIC-edge ide1 16:1695332 0 0 0 IO-APIC-level eth0 17:1510473 0 0 0 IO-APIC-level eth1 18: 129511 0 0 0 IO-APIC-level megaraid 24: 100936929 0 0 0 IO-APIC-level t4xxp Roberto Grasso Roberto Grasso Technical Account Manager Puntocontatto s.r.l. Via Alessandrini 9 20016 Pero tel e fax +39 02 38101310 Cell. 333-5253086 ___ 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 . ___ Asterisk-Users
Re: [Asterisk-Users] TE410P - Normal activity ?
Rich Adamson wrote: Is it normal for the following to occur hourly on an E1 PRI ? -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 -- B-channel 0/3 successfully restarted on span 1 -- B-channel 0/4 successfully restarted on span 1 -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 1 -- B-channel 0/7 successfully restarted on span 1 -- B-channel 0/8 successfully restarted on span 1 -- B-channel 0/9 successfully restarted on span 1 -- B-channel 0/10 successfully restarted on span 1 Yes, that is normal. However, if any calls are in progress when this happens, the calls are not dropped or impacted. Not sure why the restart code was added, but it was some time ago (maybe up to a year ago). I'd have to guess that it was added to address an issue back then and probably really isn't needed any more (but that is a 'guess'). ___ 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 . In extremely loaded PRI systems, asterisk suffers from the occasional stuck channel, usually following a bunch of frame re-tries. This restart code usually is able to clear the channel, allowing it to accept new incoming calls, so it's been a great thing to have! Regards Scott Stingel _www.evtmedia.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] Grandstream CallerID
I use SetCallerID() and it displays the number just fine on my
GrandStream phone.
Regards,
Scott Stingel
www.evtmedia.com
David Ishmael wrote:
I'm confused, should I be using SetCallerID(${CALLERIDNUM}) or
SetCIDNum(${CALLERIDNUM})? Also, I don't think it matters but I'm trying to
forward the CID coming in from the PSTN line. I know Asterisk sees the CID
because its shows up in the logs. I think I've tried just about every
combination possible and have pretty much given up hope on getting the GS
phone to display the CID.
-Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Ebneter
Sent: Tuesday, December 21, 2004 6:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream CallerID
Those suggestions were just for testing, obviously.
iax.conf
[2000]
callerid=My Name 2000
sip.conf
[2002]
type=friend
username=2002
Dialing 2002 from 2000
-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, SIP/2002|20|tT) in new stack
2000 appears on the BT100
___
hmmm... interesting. I made no changes to the config, but after this
morning when we had to restart the server my phone now actually
displayes a correct number set with the SETCIDNUM.
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Re: [Asterisk-Users] TE405P E1 coax cables with balun
Hi- Just a couple of things to check: I assume that the LED's blink RED when you first power up, meaning that the board is basically alive. Since you're E1, just checking that you've installed the 4 required jumpers on the card for E1 It's possible you need a crossover cable instead of straight through: 1 -- 4 2 -- 5 4 -- 1 5 -- 2 You can cut up a CAT5 cable to make a cross over, but note that this wiring is not the same as a CAT5 crossover. Once you make this cable, you can also use it to connect channels to each other on the card to get green lights (if your software is set up correctly). This allows you to see if the problem is outside or inside the system. Good luck! Scott Stingel www.evtmedia.com Ciro La Ferrara wrote: Hi, I am new with asterisk. I am setting a Wildcard TE405P. E1s in Italy come in on a pair of RG-59 coax cables with BNC connectors. So I need an adapter/balun http://www.allcomtlc.com/al_g703n3.htm . I have It but I am not sure that It works. I have configured my asterisk in this way: zaptel.conf span=1,1,0,ccs,hdb3,crc4 dchan=16 bchan=1-15,17-31 loadzone=it defaultzone=it zapata.conf switchtype = EuroISDN signalling = pri_cpe pridialplan = unknown context = incoming group = 2 channel = 1-15,17-31 lsmod wct4xxp51680 31 zaptel175904 64 [wct4xxp] ztcfg -vv Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. With cable plugged in, the led are turned off. What's wrong? Ciro La Ferrara ___ 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 . ___ 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] pre-installation jitters
This is not exactly leading-edge, but my customers are running 15 asterisk busy asterisk servers, all on Fedora Core 1, with kernel updates. Pretty solid Don't know about Core 3. Cheers Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: Thursday, December 02, 2004 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] pre-installation jitters On Wed, 1 Dec 2004, Samudra E. Haque wrote: I would like to build my newest server based upon Fedora Core 3, and load up asterisk. I was all set to do so.. but then I read in Asterisk Users Digest, Vol 4, Issue 404: I think you would be insane to run your production servers on Fedora Core 3. Most of the components in Fedora are experimental and not well tested. Indeed, the entire point of Fedora is to experiment. You would be much better served to run your installation on RedHat Enterprise Linux, or one of the free alternatives such as Tao Linux or White Box. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 and Dialogic LSI161SCREV2 --- Don't killme ; -)
Hi- You probably would be better off trying to sell this board on Ebay or something, although I notice even there they are not selling for very much. This board was retired a few years ago by Dialogic. The effort of writing a driver for this board in the asterisk environment would be huge - why not invest a few hundred US$ to buy a Digium board instead. Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kido noagbodji Sent: Wednesday, November 24, 2004 5:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and Dialogic LSI161SCREV2 --- Don't killme ; -) Hello all, I found a LSI161SCREV2 Dialogic board in one of my drawers, and i was wondering if by any luck, i could make some magic happen with asterisk ... If asterisk does not support it, is there any PSTN to H323 or PSTN to SIP gateway that support this dialogic card and that can be connected to an Asterisk Box? Digium, I PROMISE that I will buy my cardf rom you once my tests are conclusive ;-) Thanks, K. ___ 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] Paul Mahlers Book
From their Ebay site, I see they (Signate) charge US$23 for Global Express Mail (to be fair, only a little more than the actual cost to them). Maybe you could ask them to ship it Global Priority Mail instead (about 4-5 days to London), which has a cost of $9. If they won't do it, email me off line and I'll do it for cost. (but you'd have to pay the sales tax of 8.5% since I'm in California like they are!) Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clive Carter Sent: Tuesday, November 23, 2004 2:34 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Paul Mahlers Book Anybody know of a UK supplier of Voip Telephony with Asterisk by Paul Mahler ? I've searched on the web, and the only suppliers I can find are US based, and the postal charge is as much as the book. cheers -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ 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] Quick Questions - IVR=Auto Attendant?
I would say that the Auto Attendant function, used to allow people to select who they want to talk to, is a subset of IVR, which more generically allows the person calling to retrieve information or to control the operation of a computer or telephone switch. Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan Sent: Tuesday, November 23, 2004 8:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Quick Questions - IVR=Auto Attendant? Are IVR and Auto Attendant interchangeable terms? They both do the Press 1 for thing. Sales is asking me how to word it and I've always used both terms interchangeably. ___ 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] H323 Problems
Peter- I haven't retried this lately, but it worked fine in the past when I did. Be sure that you follow the instructions in the README exactly, especially the notes about which versions of pwlib and openH323 versions work with the current version of OH323 Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Landy Sent: Sunday, November 21, 2004 11:01 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H323 Problems New to Asterisk so I am sure this has been answered before. I can compile PWLIB and OpenH323 but when it comes to compiling asterisk-oh323 then I get all kinds of errors even though I have set the paths up in the source files. I can attach the errors if it is useful. I though however that someone must have gone through this exercise successfully. Any chance of someone giving me a quick how to so I can check I am doing it right? Regards Peter Landy ___ 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] Zaptel Compile Problems with 1.0 Stable
Matthew: Not sure if this is the problem, but I usually compile in this order, different from yours: zaptel libpri asterisk asterisk-addons regards, Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Tuesday, November 16, 2004 9:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Zaptel Compile Problems with 1.0 Stable Just recieved our T100P for testing PRI connectivity to Asterisk. I am using Asterisk 1.0 and libpri 1.0 and zaptel 1.0. I compiled/installed libpri first and had no errors. I compiled/installed zaptel second and had no errors. I compiled asterisk third and got the following warning: if [ -d CVS ] ! [ -f .version ]; then echo CVS-v1-0-10/06/04-19:07:31 .version; fi for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' make[1]: Nothing to be done for `all'. make[1]: Leaving directory `/usr/src/asterisk/res' make[1]: Entering directory `/usr/src/asterisk/channels' gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-decla rations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i6 86 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-10/06/04-19:07:31 \ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/li b/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run/ asterisk\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/aste risk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/ asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETEC T_MARTIN -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_P RI -DIAX_TRUNKING -DCRYPTO -fPIC -o chan_zap.o chan_zap.c chan_zap.c: In function `handle_pri_show_span': chan_zap.c:8267: warning: implicit declaration of function `pri_dump_info' gcc -shared -Xlinker -x -o chan_zap.so chan_zap.o -lpri -ltonezone This warning is causing a crash when I attempt to start asterisk with chan_zap.so loaded. Any ideas on why I'm getting this? [EMAIL PROTECTED] src]# grep -ir pri_dump_info asterisk/* zaptel/* libpri/* asterisk/channels/chan_zap.c: pri_dump_info(pris[span-1].pri); libpri/libpri.h:#define PRI_DUMP_INFO_STR libpri/libpri.h:extern char *pri_dump_info_str(struct pri *pri); libpri/pri.c:char *pri_dump_info_str(struct pri *pri) Is this possibly just a typo and that in chan_zap it should be pri_dump_info_str? Thanks, Matthew ___ 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] CDR MySQL Problem
I think the error is not the sock file error (this is just a harmless warning I believe), but rather the second failure to connect message. Make sure that you can log on yourself to MySql using the mysql command line with the same username and password that you specified in your conf file. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com - Original Message - From: Geraldo Fco. do Espírito Santo Jr. [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 12, 2004 9:21 AM Subject: [Asterisk-Users] CDR MySQL Problem Hi everyone, I am try to install a CDR using MySQL but I receive the follow message: [app_db.so] = (Database access functions for Asterisk extension logic) == Registered application 'DBget' == Registered application 'DBput' == Registered application 'DBdel' == Registered application 'DBdeltree' [cdr_addon_mysql.so] = (MySQL CDR Backend) == Parsing '/etc/asterisk/cdr_mysql.conf': Found Nov 12 13:13:41 WARNING[2438]: cdr_addon_mysql.c:330 my_load_module: MySQL database sock file not specified. Using default Nov 12 13:13:41 ERROR[2438]: cdr_addon_mysql.c:378 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. I am using a SuSE 8.0 (kernel 2.4.19), * 1.0.2 (Asterisk CVS-HEAD-11/07/04-23:06:58) and MySQL 4.1.7-standard. Here are the configuration file == Cdr_mysql.conf [global] hostname=localhost dbname=asteriskcdrdb password=test user=asteriskuser Thanks Geraldo ___ 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] BRI in the US
Brian- I was quoted (verbally) something on the order of $60 per month for a single BRI by SBC in San Francisco about 60 days ago. I thought that was high.. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Friday, November 12, 2004 1:23 PM To: 'Michael Bielicki'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] BRI in the US Check this out www.bkw.org/pri.pdf That's what SBC charges for PRI here... it's the only option I have right now. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael Bielicki Sent: Friday, November 12, 2004 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BRI in the US the horribly expensive EICON shit. Although if you just want to connect ISDN phones to asterisk you can use european ISDN phones with cards from Junghanns.net On Fri, 12 Nov 2004 14:10:07 -0600, Brian West [EMAIL PROTECTED] wrote: What cards will work with asterisk and BRI in the US? bkw ___ 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 -- Michael Bielicki ___ 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] Marconi Sys X/TE410P configuration
Hi Steve- Just wanted to make sure that you're aware of an earlier post regarding problems with Marconi, by Darren Storer I think: Here it is: Are you sure that NTL have provided you with a true Q.931 EuroISDN PRI circuit? If the circuit was supplied some time ago for use with existing equipment it may not be fully EuroISDN compliant. When connecting to Telcos that use Marconi (GPT) System X switches you must make sure that the circuit is using ISDN 110 and not ISDN 85. ISDN 85 is an older implementation of Q.931 in the UK and it does not work with Asterisk E100P or TE410P in my experience. When checking with NTL ask if your circuit is ISDN 110 or Q.931e or ETSI; these are the three names that are commonly used by UK carriers for full spec. EuroISDN PRI. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Tuesday, November 09, 2004 6:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Marconi Sys X/TE410P configuration Has anyone got a working config for a Marconi System X (Q.931) and Digium TE410P? Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ 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] Enquiry about Wildcard E100P card
Ning- I think it goes like this: TE410P - 3.3v PCI only TE405P - 5v PCI only E100P and T100P - 5v PCI only Better double check with Digium though - to be sure Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ning Zhou Sent: Tuesday, November 09, 2004 7:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Enquiry about Wildcard E100P card Hi, all I am new to the Asterisk PBX, I have a question and hope you can help. Our company want to buy a single port PCI card, Wildcard E100P. On the web of Digium it said that E100P has the same features as the TE410P card, but TE410P can only be used with 3.3 volt PCI slot. Is that the same case for the card E100P, only can be used for 3.3 volt PCI slot on the motherboard? Thank you so much! Best Regards, ning ___ 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] Analog answering machine hangs up early
Sounds like the asterisk system is not sourcing enough line current on the FXS port, and the answering machine may think the line has hung up. This is what the answering machine is supposed to do, for example, when someone picks up another phone on the same line. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Rizzo Sent: Thursday, October 28, 2004 5:12 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Analog answering machine hangs up early So, I've got an analog fax/answering machine connected to an FXS port on a TDM400p card, and everything is fine as long as I answer the phone manually. If, however, I let the answering machine pick it up, it hangs up a few seconds into playing the greeting. Don't ask why I want to use this analog machine instead of the voicemail app- it's a long and dull story. But I can't figure out why it's hanging up... any thoughts? Thanks, +j ___ 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] Re: Asterisk 1.0.2
Hi Andrew- I know you've likely described this before, but what again are the differences between the FC1 and RH9 implementations? I've always built them the same way. Also, what are the other devel and web RPM's that you've posted in that directory? Best regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew McRory Sent: Tuesday, October 26, 2004 10:20 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk 1.0.2 - Asterisk 1.0.2 rpms now available for FC1: ftp://ftp.linuxsys.com/pub/releases/FC1/asterisk-v1.0 rh9 and possibly rh73 coming later today Enjoy, -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. - http://www.linuxsys.com Located in beautiful Tallahassee, Florida Office 850-224-5737 Office 850-575-7213 Mobile 850-294-7567 ___ 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] AGI RECORD FILE BUG!
If you have a solidly re-produceable bug, suggest that you go to http://bugs.digium.com/login_page.php Sign up, and post the bug. Regards, Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Smith Sent: Tuesday, October 19, 2004 3:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] AGI RECORD FILE BUG! Importance: High I am experiencing a problem with the RECORD FILE functionality in AGI when I am doing a Record_file. After approx 20 mins + the Record_file ceases to accept escape digits and therefore records for ever or until my timeout I set. It acts like a dead application, just recording without the ability to stop. It basically does not allow you to use the escape with the DTMF string you give and for some reason it works perfectly fine at the beginning of the call and on small recordings. Please help It is consuming me, we have tried everything and read all the forums. Any ideas? Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronan de Kermadec Sent: Tuesday, 19 October 2004 8:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] About Supervised Call Transfert on GS BT100 Hi, I have a Grandstream Budge Tone 100 and i wanted to use the supervised call transfert feature but i don't find any tips for that. So there is my question : Is this feature is implemented on GS BT100 and if it is not, it is possible to implement it directly on Asterisk. Juts for your infomation, blind transfert work fine with the transfert key. Thanks a lot ! Ronan ___ 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 #== gPopper Menu ===# Delete from Gmail inbox: mailto:del|[EMAIL PROTECTED] Mark message as unread:mailto:unr|[EMAIL PROTECTED] Mark message as read: mailto:rea|[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 ___ 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 OS Choice
Yes, Fedora works fine. Debian too. (I've used both) ...and others have successfully used other flavours... See the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk Regards, Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Barnes Sent: Thursday, October 14, 2004 6:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Advice on OS Choice Hi all, I am currently trying to decide what Operating System is best to go for on a customer site. Server will only be running Asterisk / MySQL / Apache / PHP but nothing else. I have only tested Asterisk on SLES 8.1 however I do have experience with RedHat 9 as well. However SLES 8 = £599 ex vat and with so many free Linux OS's out there I am tempted to deploy something else on customer sites. RedHat 9 is no longer available so although I probably have CD's somewhere I assume that I cannot actually legally install this now for NEW installs I dont think I will get any value from support contracts so if there is anyone out there that has done some customer / 3rd party deployments could you offer some advice on what you have used and why? Also is there really any difference between using a Standard / Enterprise or Destop SUSE (Will never require more than two CPUs). Does SLES 8 bring £599 worth of enhancements to the table over SUSE Desktop or Fedora et al? Is Fedora a good choice? Sorry for the many questions but I am not particularly experienced with Linux to know what the real difference is with the umpteen versions. thanks as always for any advice Alex |Alex Barnes |SQA Engineer |URL : http://www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. Dear Friends of Ubiquity Software: As you may have noticed, Ubiquity Software began using the web domain ubiquity.com earlier this year in addition to the previously established ubiquity.net for our website and email communications to you. However, since that time, a dispute has emerged with respect to actual ownership of the ubiquity.com domain. As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net http://www.ubiquity.net/ http://www.ubiquity.net/ and via email at @ubiquity.net. However, we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ http://www.ubiquitysoftware.com/ for web and @ubiquitysoftware.com for email communications. Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. Thank you. Regards, Ubiquity Software www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ http://www.ubiquitysoftware.com/ [EMAIL PROTECTED] mailto:[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] TE405P and TE410P performance difference
Regis- I believe the cards are pretty identical, except for the support of the 3.3v PCI bus on the TE410P, and the 5v PCI on the TE405P. I've run extensive load tests on both cards in a large IVR environment, with one processor generating load to the other, and found them perform about the same. Using the TE405P might allow you a somewhat larger range of motherboards to choose from - ie. those that use the 5v PCI. Also see: http://www.voip-info.org/wiki-Asterisk+dimensioning Regards, Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Régis MARTIN Sent: Wednesday, October 13, 2004 1:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] TE405P and TE410P performance difference Hi, I checked the lists archive but I found only one answer to this question, so I ask it to have more returns. - Is there any difference in the performance of * by using the TE410P or TE405P. Does on of your have either did case studies and load testing to compare cards ? - Is there a card that is more scalable than the other. Thanks in advance for your answers. Regards, Regis ___ 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] Case studies for 120 simultaneous calls on IVR
Hi- I've run extensive load testing with both single and dual P4's and Xeon's (all at least 2.8GHz), and I've got 6 installed IVR systems of this size in various configurations. Asterisk can run 4 E1's (120 channels) in an IVR scenario, but just barely. With this many simultaneous calls, you may notice a bunch of frame retransmissions in the error logs, but these dont seem to effect the calls. There seems to be lots of call setup overhead, because I notice that shorter calls (5 seconds) seem to load the processor more heavily. For IVR only, I dont think its necessary to use the Xeon, or really dual processors for that matter - unless you're going to start doing transcoding or other processor-loading tasks. Please see my notes on the Wiki on this topic: http://www.voip-info.org/wiki-Asterisk+dimensioning Regards, Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Bielicki Sent: Friday, September 24, 2004 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Case studies for 120 simultaneous calls on IVR We use mostly dual opterons and they don't seem to notice the quad E1's on them On Fri, 24 Sep 2004 12:57:09 -0400, steve szmidt [EMAIL PROTECTED] wrote: On Friday 24 September 2004 12:46 pm, Christian Victor wrote: Hi Régis, We're going to build an IVR system with a TE405P and 4 E1. We're sure that the 120 channels will be filled by 120 simultaneous calls during peak, so we want to have the good server to manage this. We wonder a lot of things and maybe you could help us. - Are you ever build a similar system ? - Does linux use the advantage of Xeon processor ? so we must buy Xeon ? It does. Put I would prefer two single P4 boxes over one dual Xeon. BTW, Asterisk does utilize a dual processor very well. Whereas two computers offer redundancy. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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 -- Michael Bielicki ___ 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] Case studies for 120 simultaneous calls on IVR
Sorry, just dual Xeon's, not P4's! Scott M. Stingel Hi- I've run extensive load testing with both single and dual P4's and Xeon's (all at least 2.8GHz), and I've got 6 installed IVR systems of this size in various configurations. Hmm, I was under the impression that it was impossible to run dual P4 CPUs. I thought Intel programmed instruction in the cpu to not post if 2 CPUs were found. What MB are you using to run the dual P4 system? Gary ___ 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] Cheapest SIP Phone
There is an asterisk-biz list for this type of post. Asterisk-users is the non-commercial forum. Thanks! Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SeshKanuri Sent: Wednesday, September 22, 2004 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; 'Shaun Ewing' Subject: [Asterisk-Users] Cheapest SIP Phone Folks! Our Phones are cheap and they are selling well. We have no complaints so far. These phones are made by ATCOM, 2nd largest maker of VOIP gear in China. We are ATCOM's US distributors. We want to beat grandstream both at features and price. ___ 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] Wait()
That's strange - I'm using wait(10) as the first command for an extension (followed by an answer) in my dialplan and it works fine. But on this system, I have a CVS from June. Maybe you've discovered a recently-introduced bug? regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, September 20, 2004 3:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Wait() Hello List! All i would like to do is to wait 15 seconds, and then pick up the call for the voice box: [test2] exten = 39,1,Wait(15) exten = 39,2,Answer() exten = 39,3,Voicemail(99) exten = 39,4,Hangup() However, it does not pick up the call at all. I played around a bit, and found out that if the seconds are 9, then it wont pick it up anymore.Any idea why? Wait(9) work without problems, Wait(10) doesnt! :/ Any idea? Thanks, Mario ___ 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 and Red Hat 9
Thank you for all of the replies. I would like to build a PBX with a 16 channel pri and 36 phones. What kind of processor and memory should I look at? There are many references to system sizing, called dimensioning on the asterisk Wiki. The answer to your question depends on several factors, including the number of channels, how many will be transcoding ,and other things. Here's the link: http://www.voip-info.org/tiki-index.php?page=Asterisk Also, many readers on here would prefer that you post your messages in plain text, not HTML. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ 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 forum created
Yet another forum seems quite unnecessary to me, but I added a post there referring people to the Wiki. Scott Stingel www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Keating Sent: Friday, September 17, 2004 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk forum created I recently created an Asterisk forum within TMC's popular VoIP forums for everyone to use. http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15 http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15 Tom Keating TMC Labs http://blog.tmcnet.com/blog/tom-keating (My VoIP Blog) ___ 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 with E1 configuration
zaptel.conf looks good - you may require a ,crc4 at the end of the span line, depending on your provider. I have loadzone=us in mine as well. Change appropriately for yours. zapata.conf also looks good. I would also add the following, before the channel declaration: immediate=no pridialplan=unknown usecallerid=yes Setting immediate=no allows your calls to be answered and routed according to the DID entries you make in your extensions.conf file (the dialplan). Setting it to yes would cause the s extension to be used instead. So you need entries in the dialplan for each DID, under the context [default] which you have defined in zapata. That will take care of inbound calls. Outbound calls: Since you've defined group 1 as including all channels of your PRI, you can use the Dial command and use a g1 instead of a specific Zap channel, to allow asterisk to choose an available channel. All of this is covered on the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk Good luck with your project! Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of HengWee Chin Sent: Thursday, September 16, 2004 4:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help with E1 configuration Hi, I currently have a E100P card installed on my machine and the E1 subscription will be activated pretty soon. However, I have no idea how to configure asterisk to make inbound and outbound call using the E1. Especially for extensions.conf. Below is the configuration I used for zaptel.conf and zapata.conf. Is it possible if someone can verify if the configuration for zaptel and zapata is correct? zaptel.conf --- span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 zapata.conf --- switchtype=euroisdn signalling=pri_cpe group=1 context=default channel=1-15,17-31 I have 1 block of 10 DID numbers that will be subscribed together with E1. I am not able to find any sample for the extensions.conf to do inbound and outbound call. Is it possible for someone could post a sample of how the configuration would look like. Any setting missing for callerid support? PS: I already have an existing asterisk system running on analog ports. This is just an upgrade. Thanks in advanced. Regards, Chin _ Fast. Clear. Easy. The new MSN Search. http://search.msn.com.sg/ ___ 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] Results of 13 month study on reducingtelemarketing calls
Steve- That's an interesting/amusing story! The only thing I would worry about is using the Zapateller SIT tone as the first thing whenever there's no caller ID. In many places (like here in California), a good percentage of people have caller ID blocking on outbound calls from their home phones (something like 35% I think). I would worry that you might be losing a lot of legitimate calls (that you'd like to receive) from people who would give up forever when getting the SIT tone. In my area, I'd probably instead put the prompt that requests that they enter their number if blocked. But, everybody has a different situation! I thought the information you gathered was interesting in any case! Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Murphy Sent: Wednesday, September 15, 2004 8:17 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Results of 13 month study on reducingtelemarketing calls Hello-- I've been playing with the privacy options on my home/home-office system since August last year, and have some results, gleaned from my CDR records, which over the last 13 months, number a total of 8672, which includes incoming, as well as outgoing calls. Before I start spitting out numbers, let me note that with the current setup, I haven't had to tell a single telemarketer anything in the last, well, I don't know. I don't think I've had to talk with one all this year. So, something I'm doing is working. Let's see if we can figure out what it is. My line characteristics: I have two phone lines in rural Wyoming. One for business, one for home. Both are listed in phone book. I tried to unlist the business number. Because I live in the country, the phone numbers are listed in the wrong town. Boundaries of prefix areas sometimes defy logic. I's a long distance call to the nearest town. As far as areas go, I am a in farily backwater, remote location. I'd expect my call volume to be fairly low. Home line: I usually have 4 kids in the age range to get calls. My wife gets the gross majority of the calls. 1. NATIONAL DO NOT CALL LIST Before I go any further, let me state I signed up all my numbers as soon as the list opened. This has had a definite impact in reducing unwanted calls. I did not remove my name from the list as a control measure. I'll let someone else do that dirty work. The NO-CALL list, tho, is not completely affective, though. Charities, political parties, government recruiters, those with remote pre-existing business relationships are still clear to call. 2. ZAPATELLER. The Asterisk Zapateller application, which plays the SIT (Special Information Tones) (the dah-dee-die tone, usually followed with the female voice, The number you have dialed...) to those with no caller id, is the first app run on incoming calls on both my business and home lines. Humans usually do not react immediately and slam down the receiver when they hear these. But autodialers can. And according to my stats, they do, with reaction times varying from 1 to 3 seconds total call length. Total number of calls ending in Zapateller: 40 By Context: homeline: 23 workline: 17 In the above, over the last 13 months, 40 calls ended in the Zapateller application. It takes pretty quick reflexes for this to happen. Reflexes that only a telemarketer or his machines can develop. 3. PrivacyManager The PrivacyManager application gets run after the Zapateller on just my home line. Its function is to require that an anonymous caller enter some sort of callerid. My system will accept just about anything, but it does react strongly if the caller enters my own phone number. If a call ends in this application, it is either because they hang up at this time, or have severe physical impediments that render them unable to dial a ten-digit number. While it is impossible to tell whether people we know just can't handle this hurdle, and hang up, or a telemarketer can see the writing on the wall, and does the same, here are the numbers: Total number of calls ending in PrivacyManager: 38 By Context: homeline: 38 3. MENUS When I started, I didn't really consider that presenting the calling party with a set of choices (as to whom to talk to) as a possible telemarketing deterrent. But, months of watching the system in action has led me to he conclusion that is exactly that. Not all telemarketers call from unlisted numbers. If I'm not getting their call, and neither Zapateller nor PrivacyManager affects them, then I'm doing something right in my menus. They are hanging up during the introductions (the s priority) in my menus. Who? How many? There are many reasons why legitimate as well as telemarketers will hang up after they dial you. Wrong number, confusion, wrong selections, mind change, etc. all serve as possible reasons. Total number of incoming
RE: [Asterisk-Users] SIP Options
Could you please post this on asterisk-biz instead? Each email here is sent to over 8000 people, surely this is not a suitable place to conduct the poll you're looking for. Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jake Thompson Sent: Wednesday, September 15, 2004 8:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Options Hi All, I have been reading through the list quite a bit, and I am going to post this more as a poll than anything else. I am working on setting up a very small business with something that resembles a professional voice system. My idea is to use Asterisk with a SIP provider and SIP clients. I currently have a Vonage account already. So adding the 9.99 a month Soft Phone would be easy. However, there seems to be contriversy on weather or not this is a stable solution. The BroadVoice BYOD 100 Minute package will probably work well too since almost all calls will be incoming. Otherwise I am not too sure what else sounds good. If I could get opionions and suggestions it would be greatly appreciated. Thnaks, Jake Thompson ___ 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] TN405P running but with errors
It's normal, in fact I use it to be sure that everything's ok, since I think it will not occur unless we have no alarms on the spans! Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Victor Sent: Sunday, September 12, 2004 9:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TN405P running but with errors Hello! I am trying to install a TN405P on a P4-3GHz-HT machine running Debian Sarge with kernel 2.4.27. When I start Asterisk in -c mode it always shows == D-Channel on span 1 up == Restart on requested on entire span 1 == D-Channel on span 3 up == D-Channel on span 2 up == Restart on requested on entire span 3 == Restart on requested on entire span 2 == D-Channel on span 4 up == Restart on requested on entire span 4 -- B-channel 1 successfully restarted on span 2 -- B-channel 1 successfully restarted on span 3 -- B-channel 1 successfully restarted on span 4 -- B-channel 1 successfully restarted on span 1 . . . -- B-channel 31 successfully restarted on span 1 -- B-channel 31 successfully restarted on span 2 -- B-channel 31 successfully restarted on span 3 -- B-channel 31 successfully restarted on span 4 Once all B-channels have been restarted there seems to be no more problems. Is this normal behaviour? I saw the above messages in some posting concerning misconfigurations and now I am afraid to put the machine to production. Once I have the machine up I will gladly share my experience on using * in heavy-load enviroment. Thanks for your help Christian My /etc/zaptel.conf looks like this: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone = nl defaultzone=nl My carrier told me to use CAS signalling but then I get dchan up/dchan down messages every second and nothng works. When I use CCS inbound/outbound calls work fine. Mybe my carrier just lacks knowledge about their own equipment. ;-) My /etc/asterisk/zapate.conf is like: switchtype=euroisdn signalling=pri_cpe pridialplan=local ;overlapdial=yes ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no ;callwaiting=yes ;restrictcid=no ;usecallingpres=yes ;callwaitingcallerid=yes ;threewaycalling=yes ;transfer=yes ;cancallforward=yes ;callreturn=yes echocancel=no echocancelwhenbridged=no ;echotraining=yes ;relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 ;callgroup=1 ;pickupgroup=1 immediate=no ;callerid=2564286000 ;amaflags=default ;accountcode=lss0101 ;adsi=yes ;busydetect=yes ;busycount=4 ;callprogress=yes ;musiconhold=default ;idledial=6999 ;[EMAIL PROTECTED] ;minunused=2 ;minidle=1 ;jitterbuffers=4 channel = 1-15,17-31 channel = 32-46,48-62 channel = 63-77,79-93 channel = 94-108,110-124 ___ 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] Suggested Motherboard for TE410P
Hi Adam- I'm wondering if the TE405P might be a better choice, since it's 5 volt PCI and may allow you to consider a wider selection of motherboards. Sounds like you may not need the latest and fastest motherboards, which often use the 64bit 3.3v slots as you've probably found. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Friday, September 10, 2004 7:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Suggested Motherboard for TE410P Hi all, I'm looking for a new system which will use the TE410P. Originally I was going to use a dual Athlon MP system, but my supplier tells me these are being phased out now, and so will be difficult to find replacement parts later. So, I am looking for suggestions of suitable motherboards with 3.3V PCI slots for the following CPU types (in order of my personal preference) AMD Opteron 1xx Series AMD Athlon 64 Intel whatever... I don't really know the Intel range, since I don't like them much and never use them, but it looks like I might be running out of options. Overall, I don't really need a lot of power, the system will have upto 20 channels via zap devices, with up to 20 channels a-law sip devices. It may need to transcode to gsm or something for about 5 channels max, over IAX2 at any one time. From what I understand, the worst case would be having to transcode from alaw to g729, which hopefully will not be required. PS, in case you are wondering, I (and my supplier) have spent hours looking at different motherboard specs, and so far haven't been able to find anything suitable (except a dual opteron motherboard and just using a single CPU). Thanks, Adam ___ 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] Dell PowerEdge 750 rackmount
Hi Angel- Had trouble getting Dell's in Portugal, however customer can get HP Proliant DL320's. I had one shipped to me here, and ran it through some load tests. Seems fine. Thanks for responding! Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angel Gomez Sent: Thursday, September 02, 2004 10:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell PowerEdge 750 rackmount Hi Scott. I have used servers from advansor, one with a 2 Xeon cpus, 2 nics, hw raid and a te405p card, and another with 1 P4 cpu and 1 t100p, both working veri well. The only bad thing is that advansor site has an Altigen add ;-p Scott Stingel wrote: Hi- I have an upcoming order for a bunch of asterisk boxes, and I'm considering using an assembled package for the server, instead of building them from components as I usually do. Does anyone have experience with the Dell PowerEdge 750 server, or any other 1U rackmount server for use with asterisk? Thanks in advance Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ 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] Leaving messages on answering machines (no its notspam)
Answering machine detection is usually accomplished by analysing the timing of the voice energy in the initial answer period. People usually answer by saying: Hello, Frank Giwerski, Pencil sharpening department, or something fairly short, whereas answering messages are usually longer. So, I think the usualy method is to have the software listen to the voice energy for some initial period until there's a pause, and decide based on the duration of this energy whether it's a human or machine. But listening for a beep, although less efficient maybe, might work too! Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clayton Smith Sent: Thursday, September 02, 2004 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Leaving messages on answering machines (no its notspam) Hey there I'm trying to get asterisk to leave messages on answering machines So i have a pretty cool php notifying script (it notifys, it doesn't spam!!) to phones and cellphones Now all is fine if a human picks up, but if an answering machine picks up, well the script plays, but only the ending is recorded So really, the tricky part is knowing WHEN to leave a message Now to the best of my knowledge, there is no way to tell when an answering machine picks it (be it the sprint cellphone operator, or a home owned cellphone), but i was thinking... I could play my script using an EAGI script So i get extensions to run an EAGI script, that then manages everything, So when the call is picked up, relay the message, but if a high pitched beep is detected (via the EAGI script), repeat the message from scratch Now I'm no expert on asterisk, and i can see that this method could be a little buggy, so I'm wondering if there are any suggestions or if there is a better way to leaving messages on answering machines Any help or suggestions will be greatly appreciated Thanks ___ 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] SMS Asterisk - an explanation
Hi Julian- I was using a BT BRI line, with caller ID option enabled. Also, I had to send 1470 before the call because my customer had blocked his outgoing number on this line. So I'm certain that it works on BRI. BT says in their SIN document (Supplier's Information Note), number 413, that analogue lines and both ISDN 2e and 30e can provide this service. See paragraph 3.2 of this document. Here's the link: http://www.sinet.bt.com Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, September 01, 2004 12:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SMS Asterisk - an explanation 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
RE: [Asterisk-Users] Harddisk noise on TE410P
Claus- This is a problem that interests me, as I'm about to deploy TEN of these at a customer site, all with TE410P's. I'm currently load testing one Proliant box (3GHz P4 processor) looping 59 calls out to 59 calls in (leaving one channel open) - ie: lots of load. While I'm doing this, I call in from another asterisk box over IAX, route this call out over a TE410 channel and back in, and listen to a prompt. I don't hear any unusual noise, and the box is performing well otherwise. Please supply more detail: What kind of disk, which Linux distro - and, what is the noise you're hearing? Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup Sent: Tuesday, August 31, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Harddisk noise on TE410P Hi, I have this strange problem I need some help with.. It appears that I have harddisk noise captured by a Digium TE410P card (Same problem on 2 identical machines..) The machines are two Compaq Proliant DL320 G3's... Does anyone else have this problem.. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ 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] Harddisk noise on TE410P
Claus: One difference is that I'm using the slower ATA disk, not the SCSI. Is the noise rhythmic (periodic) or constant? If periodic, what is the time between noise bursts? Do you hear the noise throughout a call, or just occasionally? Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup Sent: Tuesday, August 31, 2004 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Harddisk noise on TE410P Hi there, The disks are SCSI Raid hotswap disks 1 RPM, P4 2.8 gig CPU, 1 Gig. of ram., and the server is running Red Hat 9.0. The sound is just like hearing a disk just muffled (sounds like strange static).. If you have a number I can call you at then you can hear it yourself. Kind Regards Claus Futtrup This message is for the designated recipient only and may contain privileged or confidential information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited. - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: 'Claus Futtrup' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 5:50 PM Subject: RE: [Asterisk-Users] Harddisk noise on TE410P Claus- This is a problem that interests me, as I'm about to deploy TEN of these at a customer site, all with TE410P's. I'm currently load testing one Proliant box (3GHz P4 processor) looping 59 calls out to 59 calls in (leaving one channel open) - ie: lots of load. While I'm doing this, I call in from another asterisk box over IAX, route this call out over a TE410 channel and back in, and listen to a prompt. I don't hear any unusual noise, and the box is performing well otherwise. Please supply more detail: What kind of disk, which Linux distro - and, what is the noise you're hearing? Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup Sent: Tuesday, August 31, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Harddisk noise on TE410P Hi, I have this strange problem I need some help with.. It appears that I have harddisk noise captured by a Digium TE410P card (Same problem on 2 identical machines..) The machines are two Compaq Proliant DL320 G3's... Does anyone else have this problem.. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ 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 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ 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] SMS Asterisk - an explanation
Maxim- This will not work through a FWD DID as you suggest. BT requires each telephone number to be registered in order to receive SMS messages. You need a either an analogue, BRI, or PRI line that terminates in your asterisk box directly. The way a line gets registered is that you must initiate a special SMS message on the asterisk box from the number you are registering (see the details on the asterisk Wiki page, under the SMS command) Once you have registered a number, you can send an SMS text message from a UK mobile to that number, and the asterisk box will receive it, assuming that you've defined SMS handling for that number in your dialplan. IMPORTANT NOTE: As of two weeks ago when I tested this, BT is only accepting SMS's from Vodafone mobiles - O2 and Orange do not work yet. (not sure about T-Mobile) You can send messages to all carriers however. This is expected to change in the next couple months when BT will accept SMS's from all carriers. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxim Litnitsky Sent: Tuesday, August 31, 2004 5:19 PM calluk.com gives free 0870 DIDs. I registered my 0870 to FWD account, and FWD passes all to my * box. When I send SMS to my 0870 DID, * shows nothing and I get SMS error. ___ ___ 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] No audio on PRI channel answered by Playback() orMeetMe()
You should be able to hear the audio - a sound card is not involved. Try inserting an answer command in the dialplan before you try to play something. Like Answer Wait (if you want) Playback Hangup Should work (using the proper dialplan commands) Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Larry Shields Sent: Friday, August 27, 2004 9:20 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe() Does Asterisk need a sound card or functional Console/dsp to answer inbound DID number from PRI and playback .gsm files? I can call from any of the SIP extensions on Asterisk and hear audio from Playback(), MeetMe(), or MOH. The problem I am having with calls from my PRI is as follows: I have an Asterisk (CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with PRI. I have a single DID number that rings in from the NEC IPX on PRI Span 1, trunk group 1. If I assign the inbound DID to ring an extension on Asterisk, ie. SIP/2000, it in fact rings and when answered I have a complete 2-way voice path. If I change the destination of the inbound DID from SIP/2000 to MeetMe() or Playback(), Asterisk will answer and I can see from the CLI the .gsm file being played but there is no playback audio heard on the calling extension. If I assign the DID to ring extension SIP/2000 and then after time-out send it to MeetMe() or Playback() it works and the caller hears the .gsm file. Any assistance in solving this problem is appreciated. What follows are two examples from what I tried in extensions.conf: This works but is not desirable: [nec_pri] ; Digital PRI from the NEAX2400 exten = 2688,1,Wait,1 exten = 2688,2,Dial(SIP/2000,3,Tr) exten = 2688,3,Wait,1 exten = 2688,4,MeetMe,|Mps exten = 2688,5,Hangup This will answer, but there is no audible playback on the channel: [nec_pri] ; Digital PRI from the NEAX2400 exten = 2688,1,Wait,3 exten = 2688,2,MeetMe,|Mps exten = 2688,3,Hangup This is what is displayed from the CLI while the calling station is connected via PRI: -- Accepting call from '2502' to '2688' on channel 0/4, span 1 -- Executing Wait(Zap/4-1, 3) in new stack -- Executing MeetMe(Zap/4-1, |Mps) in new stack -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Executing Hangup(Zap/4-1, ) in new stack == Spawn extension (nec_pri, 2688, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' MDBRIDGE*CLI Thank you, --LJ ___ 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] YAAN (Yet Another Asterisk Newbie)
You'll find the following web site to have a huge amount of information (too much really!) http://www.voip-info.org/tiki-index.php?page=Asterisk Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Covert (Sailtech) Sent: Wednesday, August 25, 2004 7:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie) I plan to set up an Asterisk server later today or tomorrow to begin putzing and learning about it. Learn by doing... I would like to cut thru some of the confusion that such a flexible system tends to breed by quickly describing my end goal and getting some input from the 'group mind' as to the pieces I should concentrate my efforts on. We are a 5-person operation with 6 VOIP numbers an old-style POTS PBX (Vodavi Starplus 616EX) and a dozen 6-line desk phone stations. Rather than using a small bank of ATAs, we would like to use an Asterisk server to 'terminate' the VOIP lines and route them to both the Starplus desk phones and to softphones running on certain workstations. That is, a new incoming call would ring both the first unused line hooked to the Starplus and the first unused line on the softphones. So... The question is... to get that to work, what sort of hardware do I need in the Asterisk box to turn the incoming VOIP calls into a two-wire POTS input for the Starplus PBX and what is a suggested softphone we can use with Asterisk? Thank you for your time, Dave Covert, KB5GOG | Sailtech | Office 281-334-4690 | Fax 281-538-3270 | Email dave@ ___ 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] Dell PowerEdge 750 rackmount
Hi- I have an upcoming order for a bunch of asterisk boxes, and I'm considering using an assembled package for the server, instead of building them from components as I usually do. Does anyone have experience with the Dell PowerEdge 750 server, or any other 1U rackmount server for use with asterisk? Thanks in advance Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ 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] Dell PowerEdge 750 rackmount
Steven Critchfield [EMAIL PROTECTED] wrote: buying a 1u server is much better than building it as there are a lot of cooling problems to overcome You're right about that - I learned a lot about 1U cooling and low-profile fans in the last one I built. It was fun, but now they want 10 boxes, delivered fast! The big thing to look into is what PCI busses the machine supports. We were very surprised with our Dell when it came with a PCIX slot and a 66mhz 64bit slot I noticed the PowerEdge 750 seems to have one of each: 32- and 64-bit PCI's, both brought to the rear panel - nice. BUT, I can't get the Dell's fast enough for this customer, so now I'm looking at the HP Proliant DL-320. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ 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] system reboot often?
Flynn: As far as max capacity, you might want to check out: http://www.voip-info.org/wiki-Asterisk+dimensioning Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of el Flynn Sent: Tuesday, August 24, 2004 6:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] system reboot often? Steven Critchfield wrote: It isn't advisable to have many Zap cards in a machine. If you are adding analog cards, you quickly run out of PCI slots before you get very far. If you are adding T1/E1 cards, you quickly get to a point where it is too risky to have that many circuits on a single x86 PC. The whole point of X86 PCs is that they are cheap enough to put several in use when you need it instead of building one behemoth machine. This may have crept up elsewhere in the list but I thought it might be relevant in this thread: Would having 2 of the Quad T1/E1 cards in a single machine, handling about 140 ZAP - ZAP channels be ok? Or would that be overloading the box and/or asterisk? Flynn ___ 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] Dell PowerEdge 750 rackmount
Hi Craig- I'm also interested in the other fellow's question: do the newer DL 320's require the keyboard to be present to pass the power on test and boot up? Thanks Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Tuesday, August 24, 2004 7:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell PowerEdge 750 rackmount Hi Steven, We have just built an Asterisk 1U server using a HP DL320 and a TE410p card. Is working well, however we were caught out when it arrived without the combo floppy/cdrom which is an expensive 'option'. We ended up installing FC2 via PXE. Is very very noisy, even with fans set to 'low'. In fact with fans on normal speed it is louder than the rest of the machines in the server room combined. Craig - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 2:56 AM Subject: RE: [Asterisk-Users] Dell PowerEdge 750 rackmount Steven Critchfield [EMAIL PROTECTED] wrote: buying a 1u server is much better than building it as there are a lot of cooling problems to overcome You're right about that - I learned a lot about 1U cooling and low-profile fans in the last one I built. It was fun, but now they want 10 boxes, delivered fast! The big thing to look into is what PCI busses the machine supports. We were very surprised with our Dell when it came with a PCIX slot and a 66mhz 64bit slot I noticed the PowerEdge 750 seems to have one of each: 32- and 64-bit PCI's, both brought to the rear panel - nice. BUT, I can't get the Dell's fast enough for this customer, so now I'm looking at the HP Proliant DL-320. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ 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] Choosing between TE405P and TE410P
Except for the PCI bus voltages, there does not appear to be any difference. I've load-tested both extensively and they perform about the same. I did have an issue with a TE410P getting stuck, ie not responding after a re-boot (but not a power down), but that seems to have resolved itself when running the same board in another chassis, so not sure if that was a design issue or not. I think the TE405P is a slightly newer design, but I'll bet they are virtually identical. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Monday, August 23, 2004 2:25 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Choosing between TE405P and TE410P Is there anything to choose, in performance, between a TE405P and a TE410P? I understand the difference between the PCI bus voltages, and certainly don't intend to try Andrew's hacksaw operation :-). But if I choose the card first, and a compatible mobo second, does it make any difference which? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] routing telephone calls via switchboard/asterisk.
Yes, it's very likely that you can perform these IVR functions within asterisk. If the realtime switching decisions are simple, they can probably be stored in the asterisk dialplan itself. Alternatively, you could retrieve them from a DB. Have you read the background material in the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stig Thune Sent: Monday, August 23, 2004 6:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] routing telephone calls via switchboard/asterisk. I'm new to this list. Reading the asterisk handbook pdf (good work) but but still have some questions. Using Trustix 2.1 and installed Asterisk via CVS, zaptel and libpri. We have a dedicated server which is connected to our telephone company. It makes us able to call ordinary phones via VOIP using Ericsson DRG22. Would like to make people able to call me - and get a message dial 1 for Hans, 2 for Eric, 3 for Hanna. Can I set up such a recording/playback software with the asterisk system ? And how can I route the calls onto the right number ? (guessing that I need to run mysql and storing all the phonenumber, IP, etc) Regards, Stig Henning ___ 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] Performance testing of asterisk
Hi Tom- I wrote that (rudimentary) Perl script last year to simulate traffic from one system to another, although it can also be used between spans on the same machine. It's much better to have the load generated on a separate system , for obvious reasons. A couple of things: the traffic generating spans should be set up as pri_net in zapata.conf. Also, you need an E1 crossover cable from sender to receiver. This is wired as: 1 -- 4 2 -- 5 4 -- 1 5 -- 2 I didn't see the rest of the thread here, but if your configuration involves transcoding, you need to build this in to your test too. I understand that there are software VoIP load generator programs that can do just this. If you are using T1's or E1's, I hope that my script will be useful in some way, as I found that there is a call setup/teardown load too when lots of calls are handled. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Tuesday, August 17, 2004 1:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Performance testing of asterisk On Tue, 2004-08-17 at 02:17, Tom Masterson wrote: What we are finding is that things work quite well with a small number of users/agents and callers i.e 10 or less. However when we put the stuff in production (normally changes to the configurations) where they can and are hit with hundreds of callers among some 60 or 70 agents we have major failures. What we are trying to do is come up with some automated way of creating the same effect so we can test changes before they go out to the rest of the group. The generator does not have to be the same box as the test asterisk box but we need to recreate the scenarios as much as possible. Scott [forgot surname] from Emerging Technologies (search the list messages from 2003) sent the attached script to the list during a discussion about stress testing asterisk. Iirc it works by looping 2 T1/E1/PRI ports on the same box and sending calls back and forth. Think it was possible also to loop calls between 2 asterisk boxes. Hope it is helpful. Regards, Patrick ___ 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] Is Meetme a generic term?
Meet-me is a generic term, came about some time in the 70's. I think the first use was in hospitals, where people would be paged on a beeper. The operator would park the call, page the doctor, and he/she would then call a number that would meet an incoming call. Cheers Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Monday, August 16, 2004 2:35 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Is Meetme a generic term? Just a trivial question: was the term Meetme invented for Asterisk as something like a brand name for its conferencing? Or was it an existing generic term for dial-in conferencing? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Question about TE405P
No - your settings are not correct. Try something like this: ZAPTEL #E400P (or TE410P in E1 mode) setup #note, may need to add ,crc4 to end of span lines: #first quad board span=1,1,0,ccs,hdb3 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 #first quad board bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 ZAPATA: immediate=no switchtype=EuroISDN signalling=pri_net pridialplan=unknown context=incoming usecallerid=yes group=1 signalling=pri_cpe channel = 1-15,17-31 channel = 32-46,48-62 channel = 63-77,79-93 channel = 94-108,110-124 Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angel Diaz Sent: Thursday, August 12, 2004 2:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Question about TE405P Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 zapata.conf [channels] context=default switchtype=euroisdn pridialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaitingcallerid=yes language=en immediate=no channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 Do you Yahoo!? Yahoo! Mail http://us.rd.yahoo.com/mail_us/taglines/50x/*http://promotions.yahoo.com/ne w_mail/static/efficiency.html - 50x more storage than other providers! ___ 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] Question about TE405P
Sorry, remove that extraneous signalling=pri_net. You should just have the pri_cpe. -Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com No - your settings are not correct. Try something like this: ZAPTEL #E400P (or TE410P in E1 mode) setup #note, may need to add ,crc4 to end of span lines: #first quad board span=1,1,0,ccs,hdb3 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 #first quad board bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 ZAPATA: immediate=no switchtype=EuroISDN signalling=pri_net pridialplan=unknown context=incoming usecallerid=yes group=1 signalling=pri_cpe channel = 1-15,17-31 channel = 32-46,48-62 channel = 63-77,79-93 channel = 94-108,110-124 Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angel Diaz Sent: Thursday, August 12, 2004 2:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Question about TE405P Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 zapata.conf [channels] context=default switchtype=euroisdn pridialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaitingcallerid=yes language=en immediate=no channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 Do you Yahoo!? Yahoo! Mail http://us.rd.yahoo.com/mail_us/taglines/50x/*http://promotions.yahoo.com/ne w_mail/static/efficiency.html - 50x more storage than other providers! ___ 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] zaphfc problems...
Gary- Try turning on BRI debugging (not intense), and post it. (just the part with errors that you get during the connection). Maybe you'll see an error relating to the numbering plan or something similar. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Pigott Sent: Thursday, August 12, 2004 2:13 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] zaphfc problems... Hi Scott, I tried making those changes and it didn't make any difference. :o( Gary - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 11, 2004 5:39 PM Subject: RE: [Asterisk-Users] zaphfc problems... Have you tried changing pridialplan and prilocaldialplan to unknown and supplying the full national number (no country code). That worked for me in the UK. I didn't follow this thread fully, so apologies if someone else already answered this. Turn on BRI debugging too to find out what's happening. Regards Scott M. Stingel ___ 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] zaphfc problems...
USE:bri debug span X Where x is the span number. I'm assuming that you're using the junghann's BRI card - apologies if this assumption is not correct! Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Pigott Sent: Thursday, August 12, 2004 3:20 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] zaphfc problems... Apologies for a stupid question, but how do I turn on BRI debugging? Gary - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, August 12, 2004 11:03 AM Subject: RE: [Asterisk-Users] zaphfc problems... Gary- Try turning on BRI debugging (not intense), and post it. (just the part with errors that you get during the connection). Maybe you'll see an error relating to the numbering plan or something similar. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Pigott Sent: Thursday, August 12, 2004 2:13 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] zaphfc problems... Hi Scott, I tried making those changes and it didn't make any difference. :o( Gary - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 11, 2004 5:39 PM Subject: RE: [Asterisk-Users] zaphfc problems... Have you tried changing pridialplan and prilocaldialplan to unknown and supplying the full national number (no country code). That worked for me in the UK. I didn't follow this thread fully, so apologies if someone else already answered this. Turn on BRI debugging too to find out what's happening. Regards Scott M. Stingel ___ 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 ___ 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] BRI and E1 in same system
Hi- Anyone using the Junghann's quad BRI card and the Digium E100P in the same system? I'm having a configuration problem where I can configure the cards one at a time (with the appropriate drivers loaded) in a system, but when I try them both together, neither will work. They both work fine one at a time. Probably has something to do with the channel numbering. I've tried numbering the channels with the E1 first (which produces lots of modprobe errors), and then with the BRI span's first, which produces no modprobe errors, but doesn't work. Here is the latter configuration: ZAPTEL.CONF (excerpt): # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami # E1 definition: span=5,0,0,ccs,hdb3,crc4 #BRI's: bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 #E1: bchan=13-27,29-43 dchan=28 --- ZAPATA.CONF (excerpt) switchtype = euroisdn ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp ; define 4 BRI's: pridialplan = unknown prilocaldialplan = unknown echocancel = yes context=incoming group = 1 ; S/T port 1 channel = 1-2 group = 2 ; S/T port 2 channel = 4-5 group = 3 ; S/T port 3 channel = 7-8 group = 4 ; S/T port 4 channel = 10-11 ; E1 - for output to external Dialogic only ; we are pri_net in this case group = 9 pridialplan = unknown signalling=pri_net channel = 13-27,29-43 -- STARTUP MODPROBES, ETC: #following for Quad BRI system: cd /usr/src/bri/bri-stuff.0.1.0-RC2g/qozap modprobe -v zaptel /var/log/asterisk/modprobe.log sleep 1 insmod -v qozap.o /var/log/asterisk/modprobe.log sleep 1 # following for single E1 system modprobe -v wct1xxp /var/log/asterisk/modprobe.log sleep 2 ztcfg -vv /var/log/asterisk/modprobe.log sleep 3 echo /var/log/asterisk/modprobe.log --- Thanks for any help! Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ 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] BRI and E1 in same system
Klaus- Yes, that was causing the problem. Assuming that the modprobe of the E1 would also do a ztcfg, I removed the ztcfg shown here, and now both boards come up and work: cd /usr/src/bri/bri-stuff.0.1.0-RC2g/qozap modprobe -v zaptel /var/log/asterisk/modprobe.log insmod -v qozap.o modprobe -v wct1xxp ztcfg -vv** REMOVED THIS Normally, when I have a system with just an E1, I do a modprobe first and then a ztcfg, which gives me a reassuring list of channels. I guess the redundant ztcfg doesn't matter in this case. Thanks for your help Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter Junghanns Sent: Thursday, August 12, 2004 3:52 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] BRI and E1 in same system Hi Scott, make sure that ztcfg is only run once. Modprobing the e100p driver probably triggers this automatically. I am still investigating why qozap/zaptel becomes unhappy when ztcfg is run twice. If you want me to take a look via ssh, let me know. best regards Klaus Am Do, den 12.08.2004 schrieb Scott Stingel um 12:40: Hi- Anyone using the Junghann's quad BRI card and the Digium E100P in the same system? I'm having a configuration problem where I can configure the cards one at a time (with the appropriate drivers loaded) in a system, but when I try them both together, neither will work. They both work fine one at a time. Probably has something to do with the channel numbering. I've tried numbering the channels with the E1 first (which produces lots of modprobe errors), and then with the BRI span's first, which produces no modprobe errors, but doesn't work. Here is the latter configuration: ZAPTEL.CONF (excerpt): # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami # E1 definition: span=5,0,0,ccs,hdb3,crc4 #BRI's: bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 #E1: bchan=13-27,29-43 dchan=28 --- ZAPATA.CONF (excerpt) switchtype = euroisdn ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp ; define 4 BRI's: pridialplan = unknown prilocaldialplan = unknown echocancel = yes context=incoming group = 1 ; S/T port 1 channel = 1-2 group = 2 ; S/T port 2 channel = 4-5 group = 3 ; S/T port 3 channel = 7-8 group = 4 ; S/T port 4 channel = 10-11 ; E1 - for output to external Dialogic only ; we are pri_net in this case group = 9 pridialplan = unknown signalling=pri_net channel = 13-27,29-43 -- STARTUP MODPROBES, ETC: #following for Quad BRI system: cd /usr/src/bri/bri-stuff.0.1.0-RC2g/qozap modprobe -v zaptel /var/log/asterisk/modprobe.log sleep 1 insmod -v qozap.o /var/log/asterisk/modprobe.log sleep 1 # following for single E1 system modprobe -v wct1xxp /var/log/asterisk/modprobe.log sleep 2 ztcfg -vv /var/log/asterisk/modprobe.log sleep 3 echo /var/log/asterisk/modprobe.log --- Thanks for any help! Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ 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] zaphfc problems...
I'm using vanilla bri-stuff-0.1.0-RC2k from www.junghanns.net and everything looks right. There are no errors or warnings during startup. It seems to work correctly (well SIP and IAX2 anyway) until I try to dial out over the ISDN line. I get the following: *CLI -- Executing Dial(SIP/602-9964, Zap/g1/226581) in new stack Aug 11 11:51:08 NOTICE[245775]: app_dial.c:719 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time Have you tried changing pridialplan and prilocaldialplan to unknown and supplying the full national number (no country code). That worked for me in the UK. I didn't follow this thread fully, so apologies if someone else already answered this. Turn on BRI debugging too to find out what's happening. Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ 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] TE410P-RED Alarm
It's difficult to give you help when you provide so little information. Please post your zaptel.conf and zapata.conf at least, and say what your 4 spans are connected to. Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SipMonster Sent: Monday, August 09, 2004 11:37 PM Hi, I'm using TE410P card with four T1 lines. I've configured all the channels in my /etc/zaptel.conf file. In zttool i'm getting OK for the Span-1 but the other three spans giving RED alarms. Pls give me your help where is the mistake. Regards Monster http://clients.rediff.com/signature/track_sig.asp ___ 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 AND 120 CONCURRENT CALLS
Hi Sebastian- My guess is that running 4 E1's with transcoding on all channels will overpower any single processor solution. I base this only on my own testing with 4 E1's running a simple IVR solution, with no transcoding. The processor (2.8 GHz Xeon) barely keeps up with this load. I have 5 systems installed that run this kind of load. I tried dual xeons and this didn't improve things much, but I was running an older 2.4 kernel. These results could be a bit pessimistic. My load tester produces a high volume of short calls. There is likely a lot of call setup teardown load contributing to the overall load. Someone should try a multiprocessor solution with 4 E1's and some transcoding and see where the limit really is. Should be easy enough if you have 2 boxes available - running 4 E1's on one to generate calls to 4 E1's on the other. regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Nocetti Sent: Friday, August 06, 2004 11:51 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS E1's, only G729 and from SIP to E1 or from E1 to SIP De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de mattf Enviado el: Viernes, 06 de Agosto de 2004 03:44 p.m. Para: '[EMAIL PROTECTED]' Asunto: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS Will you have E1s? will you restrict users to 729 or will you allow other codecs? will most calls be from SIP to SIP? or SIP to E1 lines? MATT--- -Original Message- From: Sebastian Nocetti [mailto:[EMAIL PROTECTED] Sent: Friday, August 06, 2004 12:53 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is stable doing thisbecause we wanna implement it in some locations!! Thanks All!! Sebastian. ___ 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
