Re: [asterisk-users] Newbie alert: VoIP hardware
Questions: [1] Can I use oslec for echo cancellation? I'll have beefy hardware. Is echo cancellation necessary? Yes you can use oslec provided that either your distribution has a zaptel package with the oslec patch (or you build the zaptel drivers + oslec yourself) Well without echo cancelation you will probably have a number of calls that have either very bad sound quality or are simply annoying With your set i.e 3-4 lines processing requirments are minimal so you should not worry about that.We have been able to run oslec for 4 lines on a 266Mhz (no its not Ghz) powerpc embedded board with very good results Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com http://www.digital-opsis.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Repo Sent: Wednesday, May 07, 2008 8:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie alert: VoIP hardware Hello, Please forgive me for i'm not an asterisk user yet. I've done as much research as I can .. and have the following questions. I'm setting up a new office and a home office and i'm shopping for hardware. Office: 2 analog lines Hardware: TDM412B (2 FXO, 1FXO) Link: http://www.voipsupply.com/index.php?cPath=99_555_556 Cost: $303 Home: 1 analog line Hardware: TDM421B (2 FXS, 1 FXO) Link: http://www.voipsupply.com/product_info.php?products_id=3980 Cost: $300 [2] Can I get PCI express x1 cards for the same price? I'm on budget, Any other cards (sangoma? rhino?) that might work well? I'm sure these questions have been asked before.. :-) Steve ___ -- 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] Newbie alert: VoIP hardware
Steve Repo wrote: Hello, Please forgive me for i'm not an asterisk user yet. I've done as much research as I can .. and have the following questions. I'm setting up a new office and a home office and i'm shopping for hardware. Office: 2 analog lines Hardware: TDM412B (2 FXO, 1FXO) Link: http://www.voipsupply.com/index.php?cPath=99_555_556 Cost: $303 Home: 1 analog line Hardware: TDM421B (2 FXS, 1 FXO) Link: http://www.voipsupply.com/product_info.php?products_id=3980 Cost: $300 If you only have one analogue line why not just get a simple x100p card? When you use OSLEC with them they work great here in the UK. I bought my card from a USA based eBay seller. Total cost for card and shipping was about £17.00 Questions: [1] Can I use oslec for echo cancellation? I'll have beefy hardware. Is echo cancellation necessary? I would think you will always want to have EC. Whether you will need oslec or not is another matter. If the standard MG2 sounds crap, try oslec. MG2 couldn't deal with echo on my x100p. Oslec is pretty much perfect. I don't think you will need beefy hardware either. I have our Asterisk server running on a Via CN700 (1Ghz) along with lots of other applications. No troubles. Of course it is a home box and not heavily used but hey - the mobo only draws about 7W! If you want to know more about it see my sig below. There's a series of articles about setting up and building a home server with Asterisk and other bits and bobs. HTH Alan -- The way out is open! http://www.theopensourcerer.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] Newbie alert: VoIP hardware
Alan Lord wrote: If you only have one analogue line why not just get a simple x100p card? When you use OSLEC with them they work great here in the UK. I bought my card from a USA based eBay seller. Total cost for card and shipping was about £17.00 Respectfully, I don't agree. I've purchased an original clone :-P of the X100P card, on the long period they almost always have some drawbacks... Faxing have been troubling for me. Don't know if it was for the line or else, but with a Digium card I had no problem at all. No sponsoring in here, ok, but certified hardware works better, therefore it's a better investment, I think. ___ -- 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
[asterisk-users] IAX IP Trunk + GSM Codec and Noise in the Polycom IP Phone 320
Hi All; I have an IP Trunk between two asterisk box (A and B), when side A originate calls via the digium card from the fxo port, and need to talk with side B at Polycom 320, then there is a disturbance will be heared on Polcyom 320. Note that used codec for the trunk between the side A and B is GSM, while Polcyom is using alaw codec. This problem happens only when originator is coming via the PSTN in side A and calling Polycom at side B, but if FXS port in side A is calling Polycom at side B then no disturbance. In other words, the problem happens only when someone call from landline or mobile line to Asterisk in side A, and then dial the extension of Polycom connected to Asterisk in side B. To what could be the problem? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] Newbie alert: VoIP hardware
Marco wrote: Respectfully, I don't agree. I've purchased an original clone :-P of the X100P card, on the long period they almost always have some drawbacks... Faxing have been troubling for me. Don't know if it was for the line or else, but with a Digium card I had no problem at all. No sponsoring in here, ok, but certified hardware works better, therefore it's a better investment, I think. I'm just offering my experiences. I have had no problems with my x100p card since using the oslec canceller. There's a big difference between $300 and $34 for one analogue line on a home phone. Of course YMMV ;-) Alan -- The way out is open! http://www.theopensourcerer.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
[asterisk-users] cdr question
Hi, Would just like to ask about cdr, i have an asterisk and i would like to bill only outbound calls not extension to extension, when i'm looking at the CDR, i can't figure out which fields i need to filter all outbound calls only. e.g if i dial 00. or 9XX (for local pstn calls) those are billable, 100 101 or 102 (all local extensions) not billable. *97 for voicemail not billable, but still is being logged on the cdr, can i disable logging to cdr calls like that(*98,*1,etc.)? also, the time the call ended is not logged, is there a way to log that? TIA ron - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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
[asterisk-users] update DB on ringing/ catch ringing event
Hello ppl, Anyway in Asterisk to update a DB/ do some action on events like ringing. The issue is I need to be able to hangup/cancel a call, if it's ringing(decided by the admin). This is independant of the timeout that we can specify in the Dial command. If I could somehow update a DB with the channel name on ringing, it would solve my problem. I assume NVlinedetect is one way to do it, but that isn't visible anymore, more so for Asterisk 1.4 and above. Any bright ideas on this one? cheers - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] Asterisk in Production ?
On Tue, May 06, 2008 at 09:42:17PM +0200, Benoit Plessis wrote: Tzafrir Cohen a écrit : On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote: Here it is, but since the AsteriskNow release has stripped the binary i fear it won't be of much use: Is there any -debug package for asterisknow's asterisk package? On RedHat they are generated automatically. On Debian they require some extra settings, and has been present in recent Asterisk packages (the asterisk-dbg package) but not in all of the smaller modules packages. Nope, already tried this before posting but nothing like that appears on conary I looked again at http://rbuilder.rpath.com/ and searched for the package asterisk. It does seem to have a subpackage called asterisk:debuginfo. anyway, i'll be migrating on a debian asap, since i now this much better and the advantages of AsteriskNow keep reducing Off topic: That is not to say you should not try Debian ASAP ;-) To help you with that, here's a live CD: http://updates.xorcom.com/iso/ -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Tzafrir Cohen a écrit : On Tue, May 06, 2008 at 09:42:17PM +0200, Benoit Plessis wrote: Tzafrir Cohen a écrit : On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote: Here it is, but since the AsteriskNow release has stripped the binary i fear it won't be of much use: Is there any -debug package for asterisknow's asterisk package? On RedHat they are generated automatically. On Debian they require some extra settings, and has been present in recent Asterisk packages (the asterisk-dbg package) but not in all of the smaller modules packages. Nope, already tried this before posting but nothing like that appears on conary I looked again at http://rbuilder.rpath.com/ and searched for the package asterisk. It does seem to have a subpackage called asterisk:debuginfo. I'm not able to install it but i'll look further, conary is a tricky software to say the least anyway, i'll be migrating on a debian asap, since i now this much better and the advantages of AsteriskNow keep reducing Off topic: That is not to say you should not try Debian ASAP ;-) Well i tried a debian/lenny with an mISDN patched for 2.6.24 but it lead to kernel panic / server reboot after 4/5 calls on the B410p. No problem on the T220b but i need both cards ... I think i'll have to reinstall an debian/etch and either try the packaged asterisk 1.2 or manually build an 1.4 + zaptel + misdn. Everything i was looking away from when i initially choosed asteriskNow To help you with that, here's a live CD: http://updates.xorcom.com/iso/ ___ -- 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] Receptionist SNOM-360
Hi! I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls) and 15 SIP extensions. The receptionist has a SNOM-360. How many SIP accounts would you configure on that phone? Only one would be enough? Yes. One SIP account, has a limit on concurrent calls? No, not that I am aware of (on the SNOM side, certainly not on the Asterisk side). Hint: sometimes testing things yourself is both faster better than asking a simple question on a mailing lists with thousands of subscribers... I saw that the SNOM-360 can handle up to eleven SIP accounts. 12 accounts to be precise, but account != line, so you are perfectly fine with 1 account for 4 concurrent calls. Just make sure you have enough Line buttons configured on your SNOM keypad, i.e. at least 4 as otherwise your receptionist will have fun navigating between the different calls that are on hold. And do look at the call-limit setting in sip.conf. Cheers, Philipp ___ -- 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
[asterisk-users] meetme with time condition
Hi All, How can i enable time condition on meetme? below i would like to deny callers if the time is not yet the scheduled time of the conference, but it seems like its still goes to 600,2, hope anyone can help. [meet-me-test] exten = 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3) exten = 600,2,Playback(vm-goodbye) exten = 600,3,Hangup exten = 600,4,MeetMe(600||600600) regards, nhadie ___ -- 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
[asterisk-users] Setting the TOS using IPtables screws up the DSCP field
Concise summary: When I set the TOS to Minimize-Delay the DSCP field in the packet changes from Expedited Forwarding to Unknown Here are the details: Scenario 1: IpTables is not used to set the TOS This is what the packet looks like using wireshark: Internet Protocol, Src: 59.93.192.xx (59.93.192.xx), Dst: 64.62.134.yyy (64.62.134.yyy) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xb8 (DSCP 0x2e: Expedited Forwarding; ECN: 0x00) 1011 10.. = Differentiated Services Codepoint: Expedited Forwarding (0x2e) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Scenario 2: IpTables is used to set the TOS Output of $/etc/rc.d/init.d/iptables status Table: mangle Chain PREROUTING (policy ACCEPT) num target prot opt source destination Chain INPUT (policy ACCEPT) num target prot opt source destination Chain FORWARD (policy ACCEPT) num target prot opt source destination 1TOSudp -- 0.0.0.0/00.0.0.0/0 udp dpts:5060:5069 TOS set 0x10 2TOStcp -- 0.0.0.0/00.0.0.0/0 tcp dpts:5060:5069 TOS set 0x10 3TOSudp -- 0.0.0.0/00.0.0.0/0 udp dpts:1:2 TOS set 0x10 This is what the packet looks like using wireshark: Internet Protocol, Src: 59.93.192.xx (59.93.192.xx), Dst: 64.62.134.xxx (64.62.134.xxx) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xb0 (DSCP 0x2c: Unknown DSCP; ECN: 0x00) 1011 00.. = Differentiated Services Codepoint: Unknown (0x2c) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 I have no idea what am I doing wrong. Here is some reference reading I did: http://www.tucny.com/dscptos Any pointers in the right direction will be very much appreciated. Thanks for your time, Sysadmin http://www.debtconsolidationcare.com Internets First get out of debt community ___ -- 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] TDM410P driver?
Vinícius Fontes wrote: Sorry, my fault. I did a $ grep -R -i TE410P * before asking, but in the README it was listed as TE410, so no match. The TE410P and TDM410 (no P) are very different; I don't think you actually searched for the TE410P :-) Yes, wctdm24xxp is the correct driver for all of Digium's new analog cards (TDM410, TDM800P and TDM2400P, AEX800 and AEX2400). Yes, there is inconsistency in the use of the 'P' suffix, but it won't be on any new products. Steve, you are correct that in the past the part number 'TDM410P' was a TDM400P with a single FXS module on it. The part number for that product is now TDM410B, which is a TDM410 with a single FXS module on it. Unfortunately some sites persist in referring to the TDM410 as a TDM410P (VOIPSupply, for example), so confusion reigns. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] Newbie alert: VoIP hardware
On Wed, 07 May 2008 09:58:04 +0100, Alan Lord wrote: Marco wrote: Respectfully, I don't agree. I've purchased an original clone :-P of the X100P card, on the long period they almost always have some drawbacks... Faxing have been troubling for me. Don't know if it was for the line or else, but with a Digium card I had no problem at all. No sponsoring in here, ok, but certified hardware works better, therefore it's a better investment, I think. I'm just offering my experiences. I have had no problems with my x100p card since using the oslec canceller. There's a big difference between $300 and $34 for one analogue line on a home phone. Of course YMMV ;-) If you use traditional PC hardware (ie with an available PCI slot) then you can use the TDM 4xx card from Digium. I had a TDM400p and it worked well enough, much better than the X101p. If you choose embedded hardware (Soekris, Alix, WRAP, etc) then you may not be able to add a PCI card. The external interffaces are necessary, like the Linksys SPA-2000/3000 series. These also work fairly well. They have the advantage of not requiring Zaptel. Nothing about your installation suggests that a traditional PC would be beneficial. In fact, Beefy would just me more power, heat noise. Someone recently told me that the Alix 3C systems are $150 complete from www.mini-box.com. That would seem like a bargain, and ideal for your circumstance. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- 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] Setting the TOS using IPtables screws up the DSCP field
This happens because the TOS and DSCP are the same field. TOS is the first implementation of QoS on the IP header, DSCP is it's evolution and uses the same field on the IP header, you can use only one of the two at the same time. Best Regards, On Wed, May 7, 2008 at 8:59 AM, Vikas [EMAIL PROTECTED] wrote: Concise summary: When I set the TOS to Minimize-Delay the DSCP field in the packet changes from Expedited Forwarding to Unknown Here are the details: Scenario 1: IpTables is not used to set the TOS This is what the packet looks like using wireshark: Internet Protocol, Src: 59.93.192.xx (59.93.192.xx), Dst: 64.62.134.yyy (64.62.134.yyy) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xb8 (DSCP 0x2e: Expedited Forwarding; ECN: 0x00) 1011 10.. = Differentiated Services Codepoint: Expedited Forwarding (0x2e) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Scenario 2: IpTables is used to set the TOS Output of $/etc/rc.d/init.d/iptables status Table: mangle Chain PREROUTING (policy ACCEPT) num target prot opt source destination Chain INPUT (policy ACCEPT) num target prot opt source destination Chain FORWARD (policy ACCEPT) num target prot opt source destination 1TOSudp -- 0.0.0.0/00.0.0.0/0 udp dpts:5060:5069 TOS set 0x10 2TOStcp -- 0.0.0.0/00.0.0.0/0 tcp dpts:5060:5069 TOS set 0x10 3TOSudp -- 0.0.0.0/00.0.0.0/0 udp dpts:1:2 TOS set 0x10 This is what the packet looks like using wireshark: Internet Protocol, Src: 59.93.192.xx (59.93.192.xx), Dst: 64.62.134.xxx (64.62.134.xxx) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xb0 (DSCP 0x2c: Unknown DSCP; ECN: 0x00) 1011 00.. = Differentiated Services Codepoint: Unknown (0x2c) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 I have no idea what am I doing wrong. Here is some reference reading I did: http://www.tucny.com/dscptos Any pointers in the right direction will be very much appreciated. Thanks for your time, Sysadmin http://www.debtconsolidationcare.com Internets First get out of debt community ___ -- 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 -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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
[asterisk-users] Ubuntu 8.04 + Astribank
I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I can see the channel bank with lsusb, but when I tried to use zaptel_hardware, or when I try the /etc/init.d/script, they don't see my Channel Bank. I compiled the latest Zaptel 1.4.10, with Astribank's dependecies, fxload and libusb-dev. Anyone have a similiar experience ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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] better enumlookup handler
There is a enumlookup.agi that is included with FreePBX and thus trixbox, PBX in a flash, etc. etc. If you have trouble finding it let me know and I can send you it. I can;t really vouch for its quality, but I do use it and it does work... but i;m not sure how well it handles multiple results. I know it will successfully connect to systems that give multiple results, i;m just not sure if it does infact failover if the first one doesn;t work. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Brian J. Murrell [EMAIL PROTECTED] Sent: Tuesday, May 06, 2008 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] better enumlookup handler Does anyone have a better ENUM lookup handler than the built-in ENUMLOOKUP() function? The built-in function does not properly handle multiple return values such as: 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP !^\\+1866(.*)$!sip:[EMAIL PROTECTED] . 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP !^\\+1866(.*)$!sip:[EMAIL PROTECTED] . And thus does not handle roll-over should one be unavailable for whatever reason. There is this voip-info.org wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+and+multiple+ENUM +entries but the downloads that it's pointing to seem to be dead. Sure I could take to writing an AGI script and probably be done it in a few hours, but why re-invent the wheel? Thanx, b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] better enumlookup handler
7 maj 2008 kl. 04.34 skrev Brian J. Murrell: Does anyone have a better ENUM lookup handler than the built-in ENUMLOOKUP() function? The built-in function does not properly handle multiple return values such as: 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP !^\\+1866(.*)$!sip:[EMAIL PROTECTED] . 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP !^\\+1866(.*)$!sip:[EMAIL PROTECTED] . And thus does not handle roll-over should one be unavailable for whatever reason. Quoting RFC 3824: Only one SIP URI, ideally, appears in an ENUM record set for a telephone number. While it may initially seem attractive to provide multiple SIP URIs that reach the same user within ENUM, if there are multiple addresses at which a user can be contacted, considerably greater flexibility is afforded if multiple URIs are managed by a SIP location service that is identified by a single record in ENUM. Behavior for parallel and sequential forking in SIP, for example, is better managed in SIP than in a set of ENUM records. There's a long section later on in this RFC about how to make it work if you still want to have multiple SIP records... We look forward to source code improvements! /O ___ -- 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] Setting the TOS using IPtables screws up the DSCP field
Why are you trying to change the ToS from 46 (0x2e) Expedited for the RTP/RTCP packets to 16 (0x10)? I mean... these values really only need to be meaningful to yourself, your switches, your routers etc however ToS 46 (0x2e) is the standard value for RTP / RTCP as it is basically the highest priority value you can assign to something... ToS 46 is basically suppose to indicate that it trumps all other traffic and should be send before anything else (Which is a good thing for the RTP traffic) The SIP Signalling traffic is a little less important and its standard ToS value is 26 (0x1a). You also don;t need to use IPTables to set these values... Asterisk will do it for you as long as you have installed libcaps (I believe its required for it). And I don;t know what phones you are using... but your phones are probably also setting these values for you I know the Aastra phones have QoS/ToS settings under Options - Network - Type of Service -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Vikas [EMAIL PROTECTED] Sent: Wednesday, May 07, 2008 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Setting the TOS using IPtables screws up the DSCP field Concise summary: When I set the TOS to Minimize-Delay the DSCP field in the packet changes from Expedited Forwarding to Unknown Here are the details: Scenario 1: IpTables is not used to set the TOS This is what the packet looks like using wireshark: Internet Protocol, Src: 59.93.192.xx (59.93.192.xx), Dst: 64.62.134.yyy (64.62.134.yyy) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xb8 (DSCP 0x2e: Expedited Forwarding; ECN: 0x00) 1011 10.. = Differentiated Services Codepoint: Expedited Forwarding (0x2e) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Scenario 2: IpTables is used to set the TOS Output of $/etc/rc.d/init.d/iptables status Table: mangle Chain PREROUTING (policy ACCEPT) num target prot opt source destination Chain INPUT (policy ACCEPT) num target prot opt source destination Chain FORWARD (policy ACCEPT) num target prot opt source destination 1TOSudp -- 0.0.0.0/00.0.0.0/0 udp dpts:5060:5069 TOS set 0x10 2TOStcp -- 0.0.0.0/00.0.0.0/0 tcp dpts:5060:5069 TOS set 0x10 3TOSudp -- 0.0.0.0/00.0.0.0/0 udp dpts:1:2 TOS set 0x10 This is what the packet looks like using wireshark: Internet Protocol, Src: 59.93.192.xx (59.93.192.xx), Dst: 64.62.134.xxx (64.62.134.xxx) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xb0 (DSCP 0x2c: Unknown DSCP; ECN: 0x00) 1011 00.. = Differentiated Services Codepoint: Unknown (0x2c) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 I have no idea what am I doing wrong. Here is some reference reading I did: http://www.tucny.com/dscptos Any pointers in the right direction will be very much appreciated. Thanks for your time, Sysadmin http://www.debtconsolidationcare.com Internets First get out of debt community ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] reINVITE with Dial() options -- bug 0010647
Hi everyone, I've got the same problem described in http://bugs.digium.com/view.php?id=10647 (unfortunately, the bug is closed and I could not find the way to reopen it). Wiki says, When options t, T, h, H, w, W or L (with multiple arguments) are applied, Asterisk will remain in the media path, even if canreinvite=yes'' (a SIP channel option) has been specified. But in fact, if canreinvite=yes for both peers, reINVITE is always performed, no matter which Dial() options you specify. Is there a way to fix it ? What I want to do is to tell * to reinvite for a particular extension and to stay in media path for everything else. Asterisk 1.4.19.1 Thanks, Misha ___ -- 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
[asterisk-users] voice mail indicator on phone
Is there a method from the dialplan that I can turn on a voicemail indicator on a polycom phone. Like a blinking light or something. Then I would also need to turn it off. Is there a way to do that? Jerry ___ -- 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] Newbie alert: VoIP hardware
On Wed, May 7, 2008 at 8:17 AM, Michael Graves [EMAIL PROTECTED] wrote: On Wed, 07 May 2008 09:58:04 +0100, Alan Lord wrote: Marco wrote: Respectfully, I don't agree. I've purchased an original clone :-P of the X100P card, on the long period they almost always have some drawbacks... Faxing have been troubling for me. Don't know if it was for the line or else, but with a Digium card I had no problem at all. No sponsoring in here, ok, but certified hardware works better, therefore it's a better investment, I think. I'm just offering my experiences. I have had no problems with my x100p card since using the oslec canceller. There's a big difference between $300 and $34 for one analogue line on a home phone. Of course YMMV ;-) If you use traditional PC hardware (ie with an available PCI slot) then you can use the TDM 4xx card from Digium. I had a TDM400p and it worked well enough, much better than the X101p. If you choose embedded hardware (Soekris, Alix, WRAP, etc) then you may not be able to add a PCI card. The external interffaces are necessary, like the Linksys SPA-2000/3000 series. These also work fairly well. They have the advantage of not requiring Zaptel. Nothing about your installation suggests that a traditional PC would be beneficial. In fact, Beefy would just me more power, heat noise. Someone recently told me that the Alix 3C systems are $150 complete from www.mini-box.com. That would seem like a bargain, and ideal for your circumstance. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] If your budget is tight and you want a decent card (not an X100P) with room to upgrade, then check out http://www.openvox.com.cn/products.php?genre_id=25 or http://store.getvoicecards.com/index.php?cPath=66 they are the reference design that Digium used on previous cards and are very well made. You can even use their FXO/FXS modules in a real Digium card and visa versa. The page I linked to includes the Octasic SoftEcho software. Word has it that the guy responsible for these cards was a former Digium employee back when Digium was only a few people (Mark Spencer's right hand man) and he also developed the Tormenta III card for Govarion. I have seen documents and some other things that back up this information.. Thanks, Steve Totaro ___ -- 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] Newbie alert: VoIP hardware
On Wed, May 7, 2008 at 9:00 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Wed, May 7, 2008 at 8:17 AM, Michael Graves [EMAIL PROTECTED] wrote: On Wed, 07 May 2008 09:58:04 +0100, Alan Lord wrote: Marco wrote: Respectfully, I don't agree. I've purchased an original clone :-P of the X100P card, on the long period they almost always have some drawbacks... Faxing have been troubling for me. Don't know if it was for the line or else, but with a Digium card I had no problem at all. No sponsoring in here, ok, but certified hardware works better, therefore it's a better investment, I think. I'm just offering my experiences. I have had no problems with my x100p card since using the oslec canceller. There's a big difference between $300 and $34 for one analogue line on a home phone. Of course YMMV ;-) If you use traditional PC hardware (ie with an available PCI slot) then you can use the TDM 4xx card from Digium. I had a TDM400p and it worked well enough, much better than the X101p. If you choose embedded hardware (Soekris, Alix, WRAP, etc) then you may not be able to add a PCI card. The external interffaces are necessary, like the Linksys SPA-2000/3000 series. These also work fairly well. They have the advantage of not requiring Zaptel. Nothing about your installation suggests that a traditional PC would be beneficial. In fact, Beefy would just me more power, heat noise. Someone recently told me that the Alix 3C systems are $150 complete from www.mini-box.com. That would seem like a bargain, and ideal for your circumstance. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] If your budget is tight and you want a decent card (not an X100P) with room to upgrade, then check out http://www.openvox.com.cn/products.php?genre_id=25 or http://store.getvoicecards.com/index.php?cPath=66 they are the reference design that Digium used on previous cards and are very well made. You can even use their FXO/FXS modules in a real Digium card and visa versa. The page I linked to includes the Octasic SoftEcho software. Word has it that the guy responsible for these cards was a former Digium employee back when Digium was only a few people (Mark Spencer's right hand man) and he also developed the Tormenta III card for Govarion. I have seen documents and some other things that back up this information.. Thanks, Steve Totaro Martin, I hope you don't mind me blowing your cover slightly. I did not state your name but I have been recommending your products whenever possible. The reason for posting your background info is to establish credibility, but I should have probably asked you first. Let me know if you want me to stop supplying personal details. I hope hardware sales are going well. Take it easy, Steve ___ -- 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] UK BT ISDN30e PRI Problem
So a quick update on this since I haven't had any feedback... I've just grabbed the latest trunk from the digium subversion repo; I've completely cleaned out the asterisk server and rebuilt from scratch with CentOS 5, all pre-reqs have been yum installed and the whole box has been yum update'd. I've built the latest trunk of libpri with no errors... I grabbed a copy of the latest zaptel redfone source and copied the ztd-ethmf.c from there into the appropriate source dir in the zaptel code I got from digium... I added the module to the makefile and then built and installed it. This again went without issue... I have just built and installed the asterisk trunk I got from the digium svn, and I have just ftp'd all the config files I had altered back onto the server, so sip.conf, iax.conf, extensions.conf, etc; are all exactly the same as they were before... I am waiting until out of hours tonight 6pm GMT to test to see if these versions on libpri, zaptel and asterisk fix the issues; and I will update the list to reflect either my success or failure :/ Thanks guys Mike On 5/4/08, Mike Hardman [EMAIL PROTECTED] wrote: Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon connection of the E1 I get the following in pri debug span 1: Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED Now I'm sure that this line is q931 and not 921, but I cant seem to find where I configure this... My configs are as follows: Zaptel.conf: dynamic=ethmf,eth0/00:50:C2:65:D1:DC/0,31,1 dynamic=ethmf,eth0/00:50:C2:65:D1:DC/1,31,0 dynamic=ethmf,eth0/00:50:C2:65:D1:DC/2,31,0 dynamic=ethmf,eth0/00:50:C2:65:D1:DC/3,31,0 # 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 # NOTE: Most E1 use alaw codec and this must be specified. alaw=1-124 # # Global data loadzone = uk defaultzone = uk Redfone.conf: [globals] fb=192.168.1.254 port=1 server=00:0F:B5:8D:CB:95 [span1] framing=ccs encoding=hdb3 slave [span2] framing=ccs encoding=hdb3 master [span3] framing=ccs encoding=hdb3 master [span4] framing=ccs encoding=hdb3 master zapata.conf: [trunkgroups] [channels] language=en xwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no context=from-panasonic group=0 switchtype=euroisdn signalling=pri_net channel=94-108 channel=110-124 context=from-zaptel signalling=pri_cpe group=1 channel=1-15 channel=17-31 I should note that I have tried this with and without crc4 on the spans; both with identical results. Could anybody shed any light on this or point me in the right direction? I'm now not sure what I'm missing or where to go looking for it? Thanks guys and gals. Mike ___ -- 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] voice mail indicator on phone
Jerry, I'd imagine that you can achieve this through SIP Event Notify, via AGI using sipsak (www.sipsak.org) I'm doing a similar thing with Cisco phones, and it works great. Here's an example of what I pass to the phones. NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 From: sip:asterisk;tag=2427962554 To: sip:cisco Call-ID: [EMAIL PROTECTED] CSeq: 101 NOTIFY Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: sipsak voicebox Event: simple-message-summary Content-Type: application/simple-message-summary Content-Length: 22 Niles On May 7, 2008, at 8:57 AM, Jerry Geis wrote: Is there a method from the dialplan that I can turn on a voicemail indicator on a polycom phone. Like a blinking light or something. Then I would also need to turn it off. Is there a way to do that? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie alert: VoIP hardware
On Wednesday 07 May 2008 08:00:17 Steve Totaro wrote: If your budget is tight and you want a decent card (not an X100P) with room to upgrade, then check out http://www.openvox.com.cn/products.php?genre_id=25 or http://store.getvoicecards.com/index.php?cPath=66 they are the reference design that Digium used on previous cards and are very well made. You can even use their FXO/FXS modules in a real Digium card and visa versa. I believe you're misinformed. This is not a reference design; it is a clone card, plain and simple. The only reference design (see http://www.tjnet.com/solutions/pci_phone.htm) was for a single port card with no daughterboard slots. Word has it that the guy responsible for these cards was a former Digium employee back when Digium was only a few people (Mark Spencer's right hand man) and he also developed the Tormenta III card for Govarion. I have seen documents and some other things that back up this information.. That is a sore subject, as well. As best as I can tell, Martin left the company with an agreement letting him pursue a business selling the X100P (because Digium planned to stop selling that board, and there wouldn't be a conflict), and because of a miswording of the agreement, it let him clone Digium boards that he had worked on (even though they're not exclusively his designs). Note that purchasing Digium boards helps pay for full time Asterisk development, and purchasing clone boards does not pay for even a part-time Asterisk developer. -- Tilghman ___ -- 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
[asterisk-users] SLA in 1.4.18: i'm going crazy.
Hi all, i'm trying from several days to setup a SLA on my machine with some THOMSON 2030. My goal is to bind every F key to an extension (NOT a trunk). So, F1 = 201, F2 = 202, F3 = 203, and so on... I'm googled thousand of pages and many more confusing concepts are in my mind. My server uses extensions with numbering 2XX placed in context 'phones'. I set yet in sip.conf: limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes sip show peer 222 (222 is my test phone) give me... * Name : 222 Secret : Set MD5Secret: Not set Context : phones Subscr.Cont. : phones Language : it AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 100 [] Then is set in [phone] context: exten = 222,hint,SIP/222 exten = 202,hint,SIP/202 exten = 244,hint,SIP/244 ..and so on, for each pohone. CLI show hints give me: [EMAIL PROTECTED] : SIP/222 State:Idle Watchers 0 for each peer (Watchers is ever 0 for all) Someone can clarify me, in detail, what is wrong? Thanks -- PicoStreamer - the real WEB live streaming software vinz486.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] Help Please - Asterisk MYSQL interface seems to be eating data
Al Baker schrieb: Are you saying the * server does NOT TRY to re-establish the BD connection ? The MySQL Realtime driver _does_ reconnect. (Search for mysql_reconnect() in res_config_mysql.c) If NOT, what happens to you CDR records ? Same thing with cdr_addon_mysql.c - it tries to reconnect. When there is no connection it writes the CDRs to a file and as soon as it successfully reconnects stores them in the database. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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] better enumlookup handler
On Wed, 2008-05-07 at 14:26 +0200, Johansson Olle E wrote:
Quoting RFC 3824:
Only one SIP URI, ideally, appears in an ENUM record set for a
telephone number. While it may initially seem attractive to
provide multiple SIP URIs that reach the same user within ENUM,
if
there are multiple addresses at which a user can be contacted,
considerably greater flexibility is afforded if multiple URIs are
managed by a SIP location service that is identified by a single
record in ENUM.
There are several problems with that. In my use case, it's toll-free
handling by separate SIP providers being enumerated (generically -- i.e.
they return NAPTRs for any 18{00,66,88,etc.}* numbers) for all providers
registered to handle toll-free) by e164.org. I'm not sure how feasible
it is to return a single SIP location service (I take that to mean a SRV
record) in that situation given that different providers have different
formats. See from my previous e-mail, that for a given number, say,
18668823998 the following two SIP urls can be used:
sip:[EMAIL PROTECTED] .
sip:[EMAIL PROTECTED] .
I fail to see how something like that could be coded into a single
location service record.
Additionally, I'm not even sure multiple SRV records would be any
better. Where is the handling of the fact that there is(/are multiple)
SRV records for a given SIP address done and how does rollover happen
when one of them returns CONGESTION, say?
Behavior for parallel and sequential forking in
SIP, for example, is better managed in SIP than in a set of ENUM
records.
Does this imply that if there are multiple SRV records for a resource,
say:
$ORIGIN mydomain.com
_sip._udp 3600 IN SRV 10 0 5060 asterisk1
_sip._udp 3600 IN SRV 10 0 5060 asterisk2
that Dial(SIP/[EMAIL PROTECTED]) will in fact iterate over the SRV
records in the case of connection failure of one of them?
If so, I'm not sure how/if e164.org can translate their generic
toll-free NAPTR mapping into a working SRV service instead.
We look forward to source code improvements!
I didn't really intend to bash ENUMLOOKUP() but was simply looking for
something more robust. I am sure for the case of single NAPTR records,
ENUMLOOKUP() is just fine. Sure I would like it more robust, but other
solutions exist so I'm willing to exercise them.
Well my understanding is that the enumlookup AGI script that I'm looking
for does what I want (and would like ENUMLOOKUP() to do) and that's
return all of the values from a single lookup (i.e. in an array or list)
rather than calling ENUMLOOKUP() iteratively for however many objects
exist.
Even the existing single record/iterative behaviour of ENUMLOOKUP()
would not be so bad if it kept state for each caller and actually did
return the successive records found from a single lookup rather than
doing a new lookup every time, possibly getting records in a different
order than it did last time (which of course results in handing back the
same record it did last time even though the record number counter has
been incremented).
Cheers,
b.
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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
Steve Totaro schrieb: I would not run MySQL on the local box. I would simple use Asterisk's csv CDRs and then use some script to import the CSVs into a database residing on another server using some sort of script. Depending on your needs, you could probably run that during low call volume. I also think that you adapt the free queue_log to database script by Queuemetrics to do what you want on the fly. We're using a custom script for the queue_log - db import in Gemeinschaft as well. But I'm not really happy with that. You need to run such a script at least once a minute to get real-time statistics for the GUI etc. Everything could be much nicer if Asterisk wrote the queue log into the database directly. As an alternative solution you can use a named pipe. But Asterisk is not prepared to handle the broken pipe error which occurs if your script should ever fail to read from the pipe. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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] cdr question
ronald ramos schrieb: Would just like to ask about cdr, i have an asterisk and i would like to bill only outbound calls not extension to extension, when i'm looking at the CDR, i can't figure out which fields i need to filter all outbound calls only. e.g if i dial 00. or 9XX (for local pstn calls) those are billable, 100 101 or 102 (all local extensions) not billable. *97 for voicemail not billable, but still is being logged on the cdr, can i disable logging to cdr calls like that(*98,*1,etc.)? NoCDR() ? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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] Newbie alert: VoIP hardware
On Wed, May 7, 2008 at 9:40 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 07 May 2008 08:00:17 Steve Totaro wrote: If your budget is tight and you want a decent card (not an X100P) with room to upgrade, then check out http://www.openvox.com.cn/products.php?genre_id=25 or http://store.getvoicecards.com/index.php?cPath=66 they are the reference design that Digium used on previous cards and are very well made. You can even use their FXO/FXS modules in a real Digium card and visa versa. I believe you're misinformed. This is not a reference design; it is a clone card, plain and simple. The only reference design (see http://www.tjnet.com/solutions/pci_phone.htm) was for a single port card with no daughterboard slots. Word has it that the guy responsible for these cards was a former Digium employee back when Digium was only a few people (Mark Spencer's right hand man) and he also developed the Tormenta III card for Govarion. I have seen documents and some other things that back up this information.. That is a sore subject, as well. As best as I can tell, Martin left the company with an agreement letting him pursue a business selling the X100P (because Digium planned to stop selling that board, and there wouldn't be a conflict), and because of a miswording of the agreement, it let him clone Digium boards that he had worked on (even though they're not exclusively his designs). Note that purchasing Digium boards helps pay for full time Asterisk development, and purchasing clone boards does not pay for even a part-time Asterisk developer. -- Tilghman Those agreements are not enforceable beyond a certain amount of time. I think five years has been struck down by many courts due to the nature of IT. I think one or two years is generally upheld in states that favor such agreements. Overly broad non-competes are thrown out of court left and right, even if one part of the agreement is questionable, other courts will line item sections of agreements that are not generally enforceable, while keeping the rest of the agreement intact. Miswording in a legal document is bad. I guess Digium learned a lesson on that one. As I said before, some judges will throw out entire agreements based on a single mistake. Besides, I have a feeling that he was not treated well by Digium or Govarion (this is just my opinion and have nothing to back it up) except some very interesting stories. The bottom line is, the government does not really want to inhibit your ability to earn a living but they weigh that with the harm it may cause to the company the individual has made an agreement with. I am surprised there was some sort of agreement about the X100P since it was not a direct Digium product but a (possibly slightly) modified modem with Opensource drivers. Anyways, getting back to you point about support Digium, others are suggesting purchasing the X100P (modems) with special opensource drivers. I am all for supporting Digium but more interested in support Asterisk by giving it a good reputation and exposing it to large companies including CSC, The US State Dept, and Federal Data Corp (among others I cannot speak of) Bottom line, the guy has a tight budget. I have a feeling an X100P will leave a bad taste in his mouth. I am just pointing him in a direction that will help him. My allegiances are not to Digium (although I support them myself) but to the community and especially the newbies. This is the Asterisk Users list, not the Support Digium list. I thought vendor neutrality was totally acceptable here. Thanks, Steve Totaro ___ -- 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] Ubuntu 8.04 + Astribank
On Wed, May 07, 2008 at 09:20:59AM -0300, Guilherme Loch Waltrick Góes wrote: I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I can see the channel bank with lsusb, but when I tried to use zaptel_hardware, or when I try the /etc/init.d/script, they don't see my Channel Bank. I compiled the latest Zaptel 1.4.10, with Astribank's dependecies, fxload and libusb-dev. Anyone have a similiar experience ? What is the output of zaptel_hardware? Is this an Astribank with a BRI module? If so this is unfortunetly a known issue (fixed in zaptel 1.4.10.1, but too late for Ubuntu 8.04). Still, it should show up in zaptel_hardware. What is the output of lsusb? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie alert: VoIP hardware
On Wed, May 7, 2008 at 9:40 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: Note that purchasing Digium boards helps pay for full time Asterisk development, and purchasing clone boards does not pay for even a part-time Asterisk developer. -- Tilghman BTW, I am all for having payed Asterisk Developers but I think it is needless to say that Asterisk would immediately see many more free Devs or be forked (as it has for other reasons such as the dual licensing) if Digium could not continue to provide in-house development. I know that sounds harsh but it is the truth. Thanks, Steve Totaro ___ -- 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
[asterisk-users] Problem using the sip_header-function
Hi all,
I want to get the first three Via-Header of an INVITE request to commit
them into an AGI script:
In the documentation is stated, that there are two possibilities to call
this function, the first one using only one parameter for the
SIP_HEADER-function is working:
exten = 1226,n,agi,checksomething.pl|${SIP_HEADER(VIA)}
I get the first Via-header:
-
Executing [EMAIL PROTECTED]:2] AGI(SIP/1226-081d65f0,
checksomething.pl|SIP/2.0/UDP
21.1.7.1:5060;branch=z9hG4bK845e.ce916cc.0) in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/checksomething.pl
-- AGI Script checksomething.pl completed, returning 0
-
The other possibility is to add a second parameter to define, which of
the Via-header I want:
exten = 1226,n,agi,checksomething.pl|${SIP_HEADER(VIA,1)}
exten = 1226,n,agi,checksomething.pl|${SIP_HEADER(VIA,2)}
exten = 1226,n,agi,checksomething.pl|${SIP_HEADER(VIA,3)}
This does not work, I get an empty string:
-
-- Executing [EMAIL PROTECTED]:3] AGI(SIP/1226-081d65f0,
checksomething.pl|) in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/checksomething.pl
-- AGI Script checksomething.pl completed, returning 0
-- Executing [EMAIL PROTECTED]:4] AGI(SIP/1226-081d65f0,
checksomething.pl|) in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/checksomething.pl
-- AGI Script checksomething.pl completed, returning 0
-- Executing [EMAIL PROTECTED]:5] AGI(SIP/1226-081d65f0,
checksomething.pl|) in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/checksomething.pl
-- AGI Script checksomething.pl completed, returning 0
-
What am I doing wrong?
I am using Asterisk 1.4.17~dfsg-2ubuntu1
br and thanks in advance
Michael
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Re: [asterisk-users] better enumlookup handler
On Wed, 2008-05-07 at 08:21 -0400, Matt Watson wrote: There is a enumlookup.agi that is included with FreePBX and thus trixbox, PBX in a flash, etc. etc. Yeah, I had gotten that impression somewhere too. If you have trouble finding it let me know and I can send you it. If you would be so kind, I will take you up on this offer. Saves me from having to download the whole FreePBX/trixbox, etc. just to get the one script. I wonder if the asterisk project would consider hosting that script as a contrib in the distribution. I can;t really vouch for its quality, I guess a code audit will tell. :-) Although I got an impression that it was written in PHP. I'm not much of a fan of PHP. Don't really see the point for something so simple. Bash, Perl (without the overhead of PHP) or even an executable-from-C seems more appropriate for something as relatively simple. Maybe I should also take up Johansson's suggestion and fix ENUMLOOKUP. :-) but I do use it and it does work... but i;m not sure how well it handles multiple results. I will test/audit for that specifically. Thanx! b. signature.asc Description: This is a digitally signed message part ___ -- 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] Newbie alert: VoIP hardware
On Wed, May 7, 2008 at 10:22 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Wed, May 7, 2008 at 9:40 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: Note that purchasing Digium boards helps pay for full time Asterisk development, and purchasing clone boards does not pay for even a part-time Asterisk developer. -- Tilghman BTW, I am all for having payed Asterisk Developers but I think it is needless to say that Asterisk would immediately see many more free Devs or be forked (as it has for other reasons such as the dual licensing) if Digium could not continue to provide in-house development. I know that sounds harsh but it is the truth. Thanks, Steve Totaro Interesting results in Google for TDM400P TigerJet reference design. http://www.google.com/search?hl=ensafe=offclient=firefox-arls=org.mozilla:en-US:officialhs=h9Ppwst=1sa=Xoi=spellresnum=1ct=resultcd=1q=Tigerjet+Reference+design+tdm400pspell=1 Other keywords turn up much more similar results that seem to confirm that the reference design from TigerJet was used. As with anything on the internet, take it with a grain of salt but it does have enough hits to raise questions. Thanks, Steve Totaro ___ -- 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] update DB on ringing/ catch ringing event
Benjamin Jacob schrieb: Anyway in Asterisk to update a DB/ do some action on events like ringing. The issue is I need to be able to hangup/cancel a call, if it's ringing(decided by the admin). This is independant of the timeout that we can specify in the Dial command. If I could somehow update a DB with the channel name on ringing, it would solve my problem. I assume NVlinedetect is one way to do it, but that isn't visible anymore, more so for Asterisk 1.4 and above. Any bright ideas on this one? I think there is no other solution but to listen to events on the Asterisk manager interface. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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
[asterisk-users] mISDN on Debian Lenny (was: Re: Asterisk in Production ?)
Benoit Plessis wrote: Well i tried a debian/lenny with an mISDN patched for 2.6.24 Are those patches available somewhere? Pointers? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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] Receptionist SNOM-360
FaberK schrieb: I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls) and 15 SIP extensions. The receptionist has a SNOM-360. How many SIP accounts would you configure on that phone? 1 One SIP account, has a limit on concurrent calls? Of course there is _some_ kind of limit on the Snom but 4 concurrent calls should be ok. (at least for the phone, not sure about the user) I saw that the SNOM-360 can handle up to eleven SIP accounts. I think it was 12. No idea what to use 12 accounts for. I guess that's for people who don't have a PBX but connect their phone to multiple SIP providers directly. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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] Newbie alert: VoIP hardware
On Wednesday 07 May 2008 09:40:21 Steve Totaro wrote: Interesting results in Google for TDM400P TigerJet reference design. http://www.google.com/search?hl=ensafe=offclient=firefox-arls=org.mozill a:en-US:officialhs=h9Ppwst=1sa=Xoi=spellresnum=1ct=resultcd=1q=Tiger jet+Reference+design+tdm400pspell=1 Other keywords turn up much more similar results that seem to confirm that the reference design from TigerJet was used. As with anything on the internet, take it with a grain of salt but it does have enough hits to raise questions. No, it doesn't. It simply is an oft-repeated falsehood. GO to the TigerJet page, LOOK at the reference designs. They do not hide a single reference design from the web, and NONE of them are the TDM400P design. If it was a reference design, please show the world the reference design from TigerJet. There simply isn't one, and repeating it does not make it so. -- Tilghman ___ -- 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] mISDN on Debian Lenny
Philipp Kempgen a écrit : Benoit Plessis wrote: Well i tried a debian/lenny with an mISDN patched for 2.6.24 Are those patches available somewhere? Pointers? Regards, Philipp Kempgen It's a patch i got from the gentoo portage site, should be made of some mISDN commit in the git tree. but I don't recommend using them, i got two kernel panic and a hard reboot after 4/5 calls http://kambing.ui.edu/gentoo-portage/net-dialup/misdn/files/misdn-2.6.24.diff ___ -- 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] Ubuntu 8.04 + Astribank
I'm using Zaptel 1.4.10 compiled from source, it's an Astribank with 6FXS+2FXO here's the output of some commands: [EMAIL PROTECTED]:~# invoke-rc.d asterisk stop * Stopping Asterisk PBX: asterisk ...done. [EMAIL PROTECTED]:~# invoke-rc.d zaptel restart Unloading zaptel hardware drivers:. Loading zaptel framework: done. Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. xpp_usb. No functioning zap hardware found in /proc/zaptel, loading ztdummy Running ztcfg: done. [EMAIL PROTECTED]:~# lsusb Bus 005 Device 003: ID e4e4:1150 Bus 005 Device 002: ID 058f:6362 Alcor Micro Corp. Hi-Speed 21-in-1 Flash Card Reader/Writer (Internal/External) Bus 005 Device 001: ID : Bus 004 Device 001: ID : Bus 003 Device 001: ID : Bus 002 Device 001: ID : Bus 001 Device 001: ID : [EMAIL PROTECTED]:~# zaptel_hardware [EMAIL PROTECTED]:~# On Wed, May 7, 2008 at 11:18 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, May 07, 2008 at 09:20:59AM -0300, Guilherme Loch Waltrick Góes wrote: I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I can see the channel bank with lsusb, but when I tried to use zaptel_hardware, or when I try the /etc/init.d/script, they don't see my Channel Bank. I compiled the latest Zaptel 1.4.10, with Astribank's dependecies, fxload and libusb-dev. Anyone have a similiar experience ? What is the output of zaptel_hardware? Is this an Astribank with a BRI module? If so this is unfortunetly a known issue (fixed in zaptel 1.4.10.1, but too late for Ubuntu 8.04). Still, it should show up in zaptel_hardware. What is the output of lsusb? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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] Cisco to Asterisk migration
Basic process: 1) Build the A*k server so that it has tftp installed (or another box that does) 2) Build up the SIPdefault.conf and get the firmware files in place (see Cisco docs on this, plus theres loads on the wikis). 3) Test with a single phone, change its tftp server to the asterisk. Check that : a) The firmware switches to SIP b) the phone registers to A*k and all is well. Calls can be made etc... 4) Once your happy with the A*k config and I mean ***really*** happy, then add in all the configs for the other phones (I used scripts to build mine). 5) Try a few more phones manually. But eventually just update DHCP so that the tftp server option points to the A*k server. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Femi Sent: 25 April 2008 10:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Cisco to Asterisk migration Hi Guys, I have client with a Cisco 2690 call manager solution that wants to upgrade but cannot stomach the costs of continuing with Cisco The installation will go up to 100 users The client currently has about 40 Cisco phones and would like to continue with these phones with the odd Polycom I'm looking at plugging in an Asterisk box and using the existing Cisco box as a PSTN gateway only Has anyone on the list done this? Any pitfalls or tips you would like to share? Thanks Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN on Debian Lenny
Benoit Plessis wrote: Philipp Kempgen a écrit : Benoit Plessis wrote: Well i tried a debian/lenny with an mISDN patched for 2.6.24 Are those patches available somewhere? Pointers? It's a patch i got from the gentoo portage site, should be made of some mISDN commit in the git tree. but I don't recommend using them, i got two kernel panic and a hard reboot after 4/5 calls http://kambing.ui.edu/gentoo-portage/net-dialup/misdn/files/misdn-2.6.24.diff Thanks anyway. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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] VOICEMAIL OPTIONS help needed
Steve Johnson wrote: Hi everyone, We have a particular user on our Asterisk 1.4.x system who always listens to his voicemail messages via email. - Is there some way to send the voicemail ONLY to email and not retain them on the phone? - Alternatively, can the voicemail system only keep, say, just the last 10 messages (as backup in case of email delivery failure or a message getting deleted in email accidentally before it is heard), purging out the oldest when a new one is received? (If we set the option maxmsg=10 on his mailbox in voicemail.conf, I think it will stop accepting voicemails after 10 messages, not turf the oldest one and accept a new one in its place). Everyone else uses the normal voicemail options on their phones, so the solution should be just for this single user. Thanks for any suggestions. S. from voicemail.conf (1.2.24):- ; delete=yes; After notification, the voicemail is deleted from the server. [per-mailbox only] regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] VOICEMAIL OPTIONS help needed
Steve Johnson schrieb: - Is there some way to send the voicemail ONLY to email and not retain them on the phone? delete=yes in voicemail.conf I believe. Grüße, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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
[asterisk-users] Out-Going Calleriid
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten = _9.,1,Set(CALLERID(number)=01926xx)
exten = _9.,2,Dial(ZAP/1/${EXTEN:1})
Ive tried all permutations of the CALLERID (ie CALLERID(NAME) and
CALLERID(NUMBER) but it just wont work anymore.
Zapata has the following relevant settings
usecallerid=yes
hidecallerid=no
callwaiting=yes
Im Stumped
Tim
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Re: [asterisk-users] VOICEMAIL OPTIONS help needed
see voicemail.conf.sample all the options you need are documented there. maxmsg delete On Wed, May 7, 2008 at 12:49 PM, Steve Johnson [EMAIL PROTECTED] wrote: Hi everyone, We have a particular user on our Asterisk 1.4.x system who always listens to his voicemail messages via email. - Is there some way to send the voicemail ONLY to email and not retain them on the phone? - Alternatively, can the voicemail system only keep, say, just the last 10 messages (as backup in case of email delivery failure or a message getting deleted in email accidentally before it is heard), purging out the oldest when a new one is received? (If we set the option maxmsg=10 on his mailbox in voicemail.conf, I think it will stop accepting voicemails after 10 messages, not turf the oldest one and accept a new one in its place). Everyone else uses the normal voicemail options on their phones, so the solution should be just for this single user. Thanks for any suggestions. S. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to handle multiple IPs from one SIP carrier
On my SIP carrier, I register to a proxy sipconnect.dal0.cbeyond.net which ends up being 192.168.22.212 (They supply a T1 bundle) #sip show peers Name/username HostDyn Nat ACL Port Status snip Generic-8174691929/817469 192.168.22.212 N 5060 OK (41 ms) Yesterday, they had a problem with their primary server and reverted to a backup server for about 5 minutes. As chance would have it, I received a call to one of my DIDs just before and just after the switch. As you can see below, the first call was on their primary server and the Found peer finds the Generic-8174691929 peer I have set up. Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.22.212 : 5060 (NAT) Found peer 'Generic-8174691929' Found RTP audio format 0 Found RTP audio format 100 However, just after they changed to the backup service, I received the call below. Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.25.212 : 5060 (NAT) Found no matching peer or user for '192.168.25.212:5060' Found RTP audio format 0 Found RTP audio format 100 Since it was a different IP address, it found no matching peer and failed to find a valid context to send the call to. How should this be addressed in Asterisk to allow for such an incident? Bill ___ -- 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] How to handle multiple IPs from one SIP carrier
[EMAIL PROTECTED] wrote: On my SIP carrier, I register to a proxy sipconnect.dal0.cbeyond.net which ends up being 192.168.22.212 (They supply a T1 bundle) #sip show peers Name/username HostDyn Nat ACL Port Status snip Generic-8174691929/817469 192.168.22.212 N 5060 OK (41 ms) Yesterday, they had a problem with their primary server and reverted to a backup server for about 5 minutes. As chance would have it, I received a call to one of my DIDs just before and just after the switch. As you can see below, the first call was on their primary server and the Found peer finds the Generic-8174691929 peer I have set up. Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.22.212 : 5060 (NAT) Found peer 'Generic-8174691929' Found RTP audio format 0 Found RTP audio format 100 However, just after they changed to the backup service, I received the call below. Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.25.212 : 5060 (NAT) Found no matching peer or user for '192.168.25.212:5060' Found RTP audio format 0 Found RTP audio format 100 Since it was a different IP address, it found no matching peer and failed to find a valid context to send the call to. How should this be addressed in Asterisk to allow for such an incident? Bill ___ -- 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 This is why Asterisk recommends dual registration. You reg with them for out and the reg with you for in. :) ___ -- 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] Out-Going Calleriid
The leading 0 is not part of Caller*ID. Remove it.
Tim Guy wrote:
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten = _9.,1,Set(CALLERID(number)=01926xx)
exten = _9.,2,Dial(ZAP/1/${EXTEN:1})
Ive tried all permutations of the CALLERID (ie CALLERID(NAME) and
CALLERID(NUMBER) but it just wont work anymore.
Zapata has the following relevant settings
usecallerid=yes
hidecallerid=no
callwaiting=yes
Im Stumped
Tim
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Re: [asterisk-users] Ubuntu 8.04 + Astribank
On Wed, May 07, 2008 at 01:03:53PM -0300, Guilherme Loch Waltrick Góes wrote: I'm using Zaptel 1.4.10 compiled from source, it's an Astribank with 6FXS+2FXO here's the output of some commands: [EMAIL PROTECTED]:~# invoke-rc.d asterisk stop * Stopping Asterisk PBX: asterisk ...done. [EMAIL PROTECTED]:~# invoke-rc.d zaptel restart Unloading zaptel hardware drivers:. Loading zaptel framework: done. Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. xpp_usb. No functioning zap hardware found in /proc/zaptel, loading ztdummy Running ztcfg: done. [EMAIL PROTECTED]:~# lsusb Bus 005 Device 003: ID e4e4:1150 Bus 005 Device 002: ID 058f:6362 Alcor Micro Corp. Hi-Speed 21-in-1 Flash Card Reader/Writer (Internal/External) Bus 005 Device 001: ID : Bus 004 Device 001: ID : Bus 003 Device 001: ID : Bus 002 Device 001: ID : Bus 001 Device 001: ID : [EMAIL PROTECTED]:~# zaptel_hardware [EMAIL PROTECTED]:~# I think that this is because they no longer mount usbfs by default and we rely on it for some details of the perl utilities. fxload and fpgaload, OTOH, do not rely on it. Make sure you have the package fxload installed and disconnect / reconnect the Astribank (or run '/usr/share/zaptel/xpp_fxloader usb'). To mount usbfs: mount procbususb /proc/bus/usb -t usbfs (to see if it is available: 'grep usb /proc/filesystems') But I think that apart from zaptel_hardware, our other utilities do not rely on /proc/bus/usb and should work well. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOICEMAIL OPTIONS help needed
We also have a script available (on www.generationd.com) which allows a user to reply to an emailed voicemail, which then deletes the associated VM file on the asterisk box. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: May 7, 2008 3:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] VOICEMAIL OPTIONS help needed see voicemail.conf.sample all the options you need are documented there. maxmsg delete On Wed, May 7, 2008 at 12:49 PM, Steve Johnson [EMAIL PROTECTED] wrote: Hi everyone, We have a particular user on our Asterisk 1.4.x system who always listens to his voicemail messages via email. - Is there some way to send the voicemail ONLY to email and not retain them on the phone? - Alternatively, can the voicemail system only keep, say, just the last 10 messages (as backup in case of email delivery failure or a message getting deleted in email accidentally before it is heard), purging out the oldest when a new one is received? (If we set the option maxmsg=10 on his mailbox in voicemail.conf, I think it will stop accepting voicemails after 10 messages, not turf the oldest one and accept a new one in its place). Everyone else uses the normal voicemail options on their phones, so the solution should be just for this single user. Thanks for any suggestions. S. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] AGI - Choppy Sound
Hi Marcelo, Sorry, just realized I responded to you directly rather than the list. So for the record, here's the list response. _ Hi Marcelo, What format are the recordings in? Have you tried converting them to the same format? Thanks _ From: Marcelo Freitas [mailto:[EMAIL PROTECTED] Sent: Monday, May 05, 2008 6:53 PM To: Robert Norton - SophTelecom LLC; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] AGI - Choppy Sound Hi Robert, Thanks for replying and I'm glad you have an application running nicely through phpAGI ... Now answering your questions ... what's the load on the box during the times it's choppy? I was testing at night ... I mean NO simultaneous calls going through the server ... It is good machine (Dell Xeon 2.8GHZ, 1GB RAM ... cpu load 10%) ... I also notice the problem as other calls were up too ... sometimes the quality is bad ... sometimes good Are calls in general choppy during that same point or just calls going through AGI? I never had this problem with my normal menu ... and when I call and I have the problem I tried to hangup and call the other number for the other menu ... and ... no problems ... It's hard to verify ... because I call ... it's choppy ... hangup and call again ... sometimes it's choppy sometimes not ... You mention the attendant voice becomes choppy? Yes, when I call and the call is not good, and during the same call I try to talk to one atendant, I almost cannot hear her/his, but the sound for him/her is good ... That's why I don't know if it is a problem with the recordings I did, because the agent's voice is also bad Is the attendant totally outside of your AGI scripts? I'm sorry, what did you mean ? Usually what I do is ... Answer the incoming call - send to AGI - it does the logic and play some sounds - and I do and exec_goto to an context,extension,priority that has a queue setup - and from there on they answer the calls What codecs are your clients using? The incoming calls - IAX/ilbc Connection to agents - SIP/ulaw it's the same as the other menu ... Thanks, - Original Message - Subject: RE: [asterisk-users] AGI - Choppy Sound From: Robert Norton - SophTelecom LLC [EMAIL PROTECTED] Date: Mon, May 5, 2008 19:47 Hi, I take it you've looked at all the basics, what's the load on the box during the times it's choppy? Are calls in general choppy during that same point or just calls going through AGI? You mention the attendant voice becomes choppy? Is the attendant totally outside of your AGI scripts? What codecs are your clients using? I'm working on a pretty intensive phpAGI based application and even with a decent number of calls haven't had any substantial problems, more so just with load but even with substantial activity on a fairly robust box it has been fine. Thanks -Robert Norton _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcelo Freitas Sent: Monday, May 05, 2008 4:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AGI - Choppy Sound Hi folks, I'm experiencing some problems with sound through phpAGI ... What I'm trying to do is a menu, doing some database lookups and so ... But sometimes the sound become too choppy ... just sometimes .. like 1 of 5 calls ... but is a big percentage ... And I have my current menu on the dialplan that I have no problems with it ... I'm using .gsm for both but different recordings ... Does anybody has had problems like that ? Is it AGI performance problem ... even the atendant voice becomes choppy ... So strange ... Does anybody have a recommendation ? Thanks, ___ -- 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] Out-Going Calleriid
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: 07 May 2008 20:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Out-Going Calleriid The leading 0 is not part of Caller*ID. Remove it. Thanks for your mail Eric. Its not that Im afraid. Dialing my mobile I'm still getting 'Private Caller' This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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] Out-Going Calleriid
Tim Guy schrieb:
exten = _9.,1,Set(CALLERID(number)=01926xx)
exten = _9.,2,Dial(ZAP/1/${EXTEN:1})
Ive tried all permutations of the CALLERID (ie CALLERID(NAME) and
CALLERID(NUMBER) but it just wont work anymore.
Maybe CALLERID(num) works?
Grüße,
Philipp Kempgen
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Re: [asterisk-users] better enumlookup handler
Slightly off-topic: On Wed, May 07, 2008 at 10:29:47AM -0400, Brian J. Murrell wrote: I guess a code audit will tell. :-) Although I got an impression that it was written in PHP. I'm not much of a fan of PHP. Don't really see the point for something so simple. Bash, Perl (without the overhead of PHP) or even an executable-from-C seems more appropriate for something as relatively simple. Shell scripts are often very inefficient with respect to execution time. They often use subprocesses and other programs for relatively simple tasks. While running a simple bash (or better: dash) is faster than running perl or php, running a modestly complex shell script is often slower than running the same thing with perl. The shell has also relatively poor handling of bad input. If someone can sneak in '`' or such in the wrong place at the input, the results can sometimes be, well, interesting. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] better enumlookup handler
On Wed, 2008-05-07 at 22:54 +0300, Tzafrir Cohen wrote: Slightly off-topic: Yeah. On Wed, May 07, 2008 at 10:29:47AM -0400, Brian J. Murrell wrote: I guess a code audit will tell. :-) Although I got an impression that it was written in PHP. I'm not much of a fan of PHP. Don't really see the point for something so simple. Bash, Perl (without the overhead of PHP) or even an executable-from-C seems more appropriate for something as relatively simple. Shell scripts are often very inefficient with respect to execution time. Indeed. They often use subprocesses and other programs for relatively simple tasks. Agreed. While running a simple bash (or better: dash) is faster than running perl or php, running a modestly complex shell script is often slower than running the same thing with perl. Right. Execve() and library loading and so forth. Understood completely. All of that is why I did mention executable-from-C though. :-) b. signature.asc Description: This is a digitally signed message part ___ -- 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] Out-Going Calleriid
Tim Guy wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: 07 May 2008 20:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Out-Going Calleriid The leading 0 is not part of Caller*ID. Remove it. Thanks for your mail Eric. Its not that Im afraid. Dialing my mobile I'm still getting 'Private Caller' This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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 Did the type of circuit you use change? That is, were you using a digital line which passed caller-id during testing, then on the new box use an analog, or a circuit which doesn't pass caller-id? Mik ___ -- 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] better enumlookup handler
At 10:04 AM -0400 2008/5/7, Brian J. Murrell wrote:
On Wed, 2008-05-07 at 14:26 +0200, Johansson Olle E wrote:
Quoting RFC 3824:
Only one SIP URI, ideally, appears in an ENUM record set for a
telephone number. While it may initially seem attractive to
provide multiple SIP URIs that reach the same user within ENUM,
if
there are multiple addresses at which a user can be contacted,
considerably greater flexibility is afforded if multiple URIs are
managed by a SIP location service that is identified by a single
record in ENUM.
There are several problems with that. In my use case, it's toll-free
handling by separate SIP providers being enumerated (generically -- i.e.
they return NAPTRs for any 18{00,66,88,etc.}* numbers) for all providers
registered to handle toll-free) by e164.org. I'm not sure how feasible
it is to return a single SIP location service (I take that to mean a SRV
record) in that situation given that different providers have different
formats. See from my previous e-mail, that for a given number, say,
18668823998 the following two SIP urls can be used:
sip:[EMAIL PROTECTED] .
sip:[EMAIL PROTECTED] .
I fail to see how something like that could be coded into a single
location service record.
Additionally, I'm not even sure multiple SRV records would be any
better. Where is the handling of the fact that there is(/are multiple)
SRV records for a given SIP address done and how does rollover happen
when one of them returns CONGESTION, say?
Behavior for parallel and sequential forking in
SIP, for example, is better managed in SIP than in a set of ENUM
records.
Does this imply that if there are multiple SRV records for a resource,
say:
$ORIGIN mydomain.com
_sip._udp 3600 IN SRV 10 0 5060 asterisk1
_sip._udp 3600 IN SRV 10 0 5060 asterisk2
that Dial(SIP/[EMAIL PROTECTED]) will in fact iterate over the SRV
records in the case of connection failure of one of them?
If so, I'm not sure how/if e164.org can translate their generic
toll-free NAPTR mapping into a working SRV service instead.
We look forward to source code improvements!
I didn't really intend to bash ENUMLOOKUP() but was simply looking for
something more robust. I am sure for the case of single NAPTR records,
ENUMLOOKUP() is just fine. Sure I would like it more robust, but other
solutions exist so I'm willing to exercise them.
Well my understanding is that the enumlookup AGI script that I'm looking
for does what I want (and would like ENUMLOOKUP() to do) and that's
return all of the values from a single lookup (i.e. in an array or list)
rather than calling ENUMLOOKUP() iteratively for however many objects
exist.
Even the existing single record/iterative behaviour of ENUMLOOKUP()
would not be so bad if it kept state for each caller and actually did
return the successive records found from a single lookup rather than
doing a new lookup every time, possibly getting records in a different
order than it did last time (which of course results in handing back the
same record it did last time even though the record number counter has
been incremented).
Cheers,
b.
1) The ENUMLOOKUP function is currently being fixed for TRUNK.
Take a look at http://bugs.digium.com/view.php?id=8089 for the
current status. Testing would be appreciated.
2) I will generate a dialplan subroutine that will hand back an array
of SIP URIs for a given number, and I'll post it here and on the
voip-info.org wiki - that's pretty easy. I agree that one query
should result in a static and ordered set of URIs for that particular
attempt cycle.
3) Your last comment about keeping state is difficult to square
with the intent of NAPTR lookups. The point is to have dynamic NAPTR
replies in case the distant system wishes to change the inbound
behavior towards their systems, so caching that data is almost always
a Bad Idea for more than a few seconds. Creating an array of results
and then having that variable follow the caller through a very short
timeframe of cascading attempts makes sense to avoid re-ordering
confusion, but I would say that a completely new lookup to the DNS
should happen after the interval described by the last-attempted set
of responses. In other words, if we get back three SIP URIs from the
NAPTR lookup, then try each in turn until they all fail. Each
failure (depending on how your SIP timers are set) may take 20
seconds. Therefore, for that particular user session, don't do
another DNS query for 60 seconds, which is how long it takes all
three current URIs time out.
4) SRV records are an entirely different story, and unrelated to
NAPTR queries, even though it seems they are very similar. I can't
say I know precisely how SRV records are currently handled by
Asterisk, but I suspect they are not cascaded as per the RFC in the
event of failures, or load-shared as per the RFC. Can someone else
comment on this? Olle?
5) AGI
Re: [asterisk-users] Out-Going Calleriid
On Wed, May 07, 2008 at 07:46:59PM +0100, Tim Guy wrote:
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten = _9.,1,Set(CALLERID(number)=01926xx)
exten = _9.,2,Dial(ZAP/1/${EXTEN:1})
Ive tried all permutations of the CALLERID (ie CALLERID(NAME) and
CALLERID(NUMBER) but it just wont work anymore.
Zapata has the following relevant settings
usecallerid=yes
hidecallerid=no
callwaiting=yes
I presume you're running a PRI or BRI line from Virgin? If you have an
analogue line it's unlikely you have the ability to set your CLID.
If you have an ISDN variant, then you should be able to set it, but
generally (depending on how the switch is set-up) only to numbers
associated with that line. It also depends on the number of digits that
the switch expects i.e. it may just be 1926xx, or it may well be
just the xx part.
Steve
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[asterisk-users] URGENT
Hello, I have given up hope of finding a solution for my problem and I think this is my last resort. At my company I have a Trixbox box and I used freePBX to configure the pbx. They have a queue with 15 static members.. they are not online all the time.. still when ever a call comes in the queue rings ALL members without skipping unavailable ones offline ones which creats so much load on the system. I can't teach the agents to REMEMBER logging off the queue and logging in .. so Static agents is what I have to do.. now it is a pain in the butt to manually login and logout agents and I can't get the agents.conf thing work properly or work at ALL.. Is there anyway that the queue senses the offline users and logthem off or at least skip them without the need for a manual operation? Tarek Sawah Technical Advisor ___ -- 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] better enumlookup handler
On Wed, 2008-05-07 at 13:40 -0700, John Todd wrote: 1) The ENUMLOOKUP function is currently being fixed for TRUNK. Ahhh. Sweet. I wonder how difficult a backport will be. Take a look at http://bugs.digium.com/view.php?id=8089 for the current status. Testing would be appreciated. Will do. I'm afraid I don't have any way to test TRUNK however. I only have my production system and taking the phone offline just does not fly here. :-( 2) I will generate a dialplan subroutine that will hand back an array of SIP URIs for a given number, and I'll post it here and on the voip-info.org wiki - that's pretty easy. Cool. But one need not be limited to SIP of course. IAX2, and even others depending on what one might want to allow their users to do. i.e. a mailto could even be used to send a mail with a voice attachment. I agree that one query should result in a static and ordered set of URIs for that particular attempt cycle. Great. 3) Your last comment about keeping state is difficult to square with the intent of NAPTR lookups. The point is to have dynamic NAPTR replies in case the distant system wishes to change the inbound behavior towards their systems, so caching that data is almost always a Bad Idea for more than a few seconds. Ahhh. Yes. I failed to explain that part correctly. I only meant caching the data long enough to iterate through a list of ENUMLOOKUP()s as a single transaction. Clearly returning an array makes more sense. I was just trying to make a suggestion that would appease a desire to not disturb the API of ENUMLOOKUP(). Creating an array of results and then having that variable follow the caller through a very short timeframe of cascading attempts makes sense to avoid re-ordering confusion, but I would say that a completely new lookup to the DNS should happen after the interval described by the last-attempted set of responses. In other words, if we get back three SIP URIs from the NAPTR lookup, then try each in turn until they all fail. Agreed. Each failure (depending on how your SIP timers are set) may take 20 seconds. Therefore, for that particular user session, don't do another DNS query for 60 seconds, which is how long it takes all three current URIs time out. Right. I still like an array of all URIs returned from one lookup better though. 4) SRV records are an entirely different story, and unrelated to NAPTR queries, even though it seems they are very similar. Yeah, I think the spirit was there in suggesting SRV records, but the technicalities of this use case make it impossible to use SRV records. 5) AGI scripts for DNS lookups: This makes me feel like I need a shower. Agreed. But it's the shortest distance between point A and B for a price. In my installation I'm willing to pay it. Other installations might not. b. signature.asc Description: This is a digitally signed message part ___ -- 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
[asterisk-users] Realtime status feature - user feedback needed
Hello users, I had developed several patches that allows to monitor current status of queues/channels in realtime db. For example specifying realtime family channels will make asterisk to keep current channel list in realtime database engine. The same would be for queue callers and queue members (already partially available in 1.4). However I encountered a resistance from Asterisk developers, as they don't wish to accept my patches - because they don't wish to support another interface. As I read in Bug Guidelines, if enough users request this, it should make into asterisk, so I'm asking You to express Your opinion on those features. *** So, realtime status - what's this all about? Basically you get output of show channels, show queues, etc directly in Realtime table (Realtime = database engine system for Asterisk). So, Asterisk will automatically update database upon any changes of channels or queues. *** Why would You need that? In beginning I created this in order to deal with large amount of monitoring software. If there's lot of users monitoring status, some kind of cache should be put into place. With current Asterisk interfaces this would mean either inquiring current status or developing a daemon that follows up all events and collects them to keep current picture always ready. I just decided to move this layer to database engine, which deals really good with this stuff. *** Rapid development of monitoring tools What it takes to create custom monitoring tool now? AMI event listener? AJAX page that gets changes from built-in webserver? All this takes lot of time to learn and start using. Adding just few config lines in extconfig.conf would allow to automatically populate database with current status - so it's accessible easily from any programming language. All the info is just there, no need for processing or analyzing. *** Performance / Scalability Inquerying queue status means that there is lock put on queue list, all queues are traversed, information gathered and then returned. If lot of instances of monitoring software need to have this information, it's obvious that this would mean too much locks. So, as database update is thrown whenever some change is happening, it means that no additional locks are created for monitoring purposes. Transaction is sent to database engine, which keeps relatively small tables of current status. Then any number of clients can access data directly without any locking. Even 200 concurrent calls with 10 new calls per minute would still be a tiny load for MySQL. This can also be scaled by moving database to another machine, thus allowing more raw CPU usage for Asterisk. *** Development maintenance Those changes doesn't introduce any new functions in asterisk code. They utilize currently available Realtime engine which is meant for storage of configuration data. This just extends use of this engine also for status data, so maintenance of this interface should not take lot of work. *** Current patches If You are interested in using those changes right away, here are some backports for 1.4: Channels: http://ftp.iq-labs.net/realtime_channels/ Queue callers: http://ftp.iq-labs.net/realtime_queue_callers-1.4/ Queue members: work in progress, needs some refactoring and optimization to make that effective. Meetme: planned, no patches yet To use any of those patches, you will need to add backport of store/destroy to 1.4: http://ftp.iq-labs.net/realtime_store_destroy-1.4/ *** Supporting this feature If You find that those features would be good for merging into Asterisk, please write a comment in bugtracker: http://bugs.digium.com/view.php?id=12556 Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] update DB on ringing/ catch ringing event
On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: Benjamin Jacob schrieb: Anyway in Asterisk to update a DB/ do some action on events like ringing. The issue is I need to be able to hangup/cancel a call, if it's ringing(decided by the admin). This is independant of the timeout that we can specify in the Dial command. If I could somehow update a DB with the channel name on ringing, it would solve my problem. I assume NVlinedetect is one way to do it, but that isn't visible anymore, more so for Asterisk 1.4 and above. Any bright ideas on this one? I think there is no other solution but to listen to events on the Asterisk manager interface. For now, not really. You could try Realtime Channels patch I just mentioned here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211136.html This should give you up-to-date list of channels in database, so you can use SELECT * FROM channels WHERE state=Ring; to get currently ringing channels. If You find this patch useful, please add a comment to issue http://bugs.digium.com/view.php?id=12556 that you would like to see Realtime status implemented in future versions of Asterisk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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
[asterisk-users] Asterisk 1.4.20-rc1 Now Available
The Asterisk development team has released Asterisk version 1.4.20-rc2. This release is a release candidate for the upcoming official release of 1.4.20. It includes a fix for a SIP channel driver regression introduced in 1.4.20-rc1, among a number of other changes. For a full list of changes since the last release candidate, view the contents of the ChangeLog that is distributed with the release. The release candidate is available on the download site. http://downloads.digium.com/pub/telephony/asterisk Please provide release candidate testing feedback to the asterisk-dev mailing list, or the issue tracker, http://bugs.digium.com/. Thank you for your continued support of Asterisk! ___ -- 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] update DB on ringing/ catch ringing event
Atis Lezdins schrieb: On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: Benjamin Jacob schrieb: Anyway in Asterisk to update a DB/ do some action on events like ringing. The issue is I need to be able to hangup/cancel a call, if it's ringing(decided by the admin). This is independant of the timeout that we can specify in the Dial command. If I could somehow update a DB with the channel name on ringing, it would solve my problem. I assume NVlinedetect is one way to do it, but that isn't visible anymore, more so for Asterisk 1.4 and above. Any bright ideas on this one? I think there is no other solution but to listen to events on the Asterisk manager interface. For now, not really. You could try Realtime Channels patch I just mentioned here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211136.html Yeah, of course you can do almost anything with a patch. Grüße, Philipp Kempgen -- Asterisk-Tag.org 2008, May 26th/27th - http://www.asterisk-tag.org amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Melbourne Asterisk night
On Wed, 2008-05-07 at 11:44 +1000, Paul Hales wrote: Tomorrow night is the monthly Asterisk night...in melbourne (australia)... The usual stuff - get together, eat, show off tech toys. At the Pint on Punt, from 7pm. later, PaulH Love to come, but as my bike got a flat tire, won't make it in time ;-) Alas. ___ -- 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
[asterisk-users] Mediatrix 2102's
Hello all. I'm encountering an issue whereby a Mediatrix 2102 is able to register, authenticate, and place a call into an asterisk box. However, the problem happens when the asterisk box tries to proxy the call to another Mediatrix 2102, or back to the other port on the same Mediatrix 2102. No call progression, and times out trying to send a message back to the Mediatrix, or to a new one. Please find attached a wireshark packet capture of the issue. If anyone could give me an idea as to where I can start looking to solve this issue, or if they have some experience in getting Mediatrix devices talking to each other through Asterisk, it would be greatly appreciated. Thanks. wireshark-mediatrix.acp Description: application/extension-acp ___ -- 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
[asterisk-users] show CODEC in CDR
Hi, In asterisk is there a way of saving the voice codec used in the call in the CDR. I am having mostly SIP calls, with few H323 calls. I have been trying for the past 2 weeks to figure it out on my own, but with no luck. There are no channel variables that can give the current codec used in the call, or used in the channel. I need it because I must charge the clients according to the codec for the individual calls. I even tried looking in the AGI commands for asterisk, and many external AGI packages, but with no luck. Can this be done? Your help is much appreciated. Thank you and best regards, Antoine Megalla. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] update DB on ringing/ catch ringing event
On Thu, May 8, 2008 at 12:34 AM, Philipp Kempgen [EMAIL PROTECTED] wrote: Atis Lezdins schrieb: On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: Benjamin Jacob schrieb: Anyway in Asterisk to update a DB/ do some action on events like ringing. The issue is I need to be able to hangup/cancel a call, if it's ringing(decided by the admin). This is independant of the timeout that we can specify in the Dial command. If I could somehow update a DB with the channel name on ringing, it would solve my problem. I assume NVlinedetect is one way to do it, but that isn't visible anymore, more so for Asterisk 1.4 and above. Any bright ideas on this one? I think there is no other solution but to listen to events on the Asterisk manager interface. For now, not really. You could try Realtime Channels patch I just mentioned here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211136.html Yeah, of course you can do almost anything with a patch. Well, this wasn't specifically written for this requirement. I just want to add some general usage realtime status in Asterisk, and I need user support there :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] Realtime status feature - user feedback needed
Atis Lezdins schrieb: I had developed several patches that allows to monitor current status of queues/channels in realtime db. [...] +1 as long as the user can choose whether they want realtime status data in the database. *** Supporting this feature If You find that those features would be good for merging into Asterisk, please write a comment in bugtracker: http://bugs.digium.com/view.php?id=12556 Not sure if the bugtracker is the right place to write me too for a feature request type of bug. Grüße, Philipp Kempgen -- Asterisk-Tag.org 2008, May 26th/27th - http://www.asterisk-tag.org amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.20-rc1 Now Available
The subject should read Asterisk 1.4.20-rc2 Now Available On Wed, May 7, 2008 at 11:24 AM, The Asterisk Development Team [EMAIL PROTECTED] wrote: The Asterisk development team has released Asterisk version 1.4.20-rc2. This release is a release candidate for the upcoming official release of 1.4.20. It includes a fix for a SIP channel driver regression introduced in 1.4.20-rc1, among a number of other changes. For a full list of changes since the last release candidate, view the contents of the ChangeLog that is distributed with the release. The release candidate is available on the download site. http://downloads.digium.com/pub/telephony/asterisk Please provide release candidate testing feedback to the asterisk-dev mailing list, or the issue tracker, http://bugs.digium.com/. Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE:Asterisk 3rd party developed commercial software sales licensing platform
Gentlemen, Dean Collins alerted me to this thread which I had skipped over. (Thanks, Dean.) I thought I'd offer my viewpoint on the matter; please take it for what it is - just another opinion, although I hope it is an informed one. From my personal experience with buying software, licensing, and even music online, I've come to the conclusion that the best way to monetize an application or module is to make it easy for your paying customers to pay. Since thieves and hackers will always find ways around any security it is pointless to spend lots of time and money making something uncrackable, especially if that security implementation becomes onerous for your paying customers. My viewpoint is this: make it easier to do a legit install than to circumvent the security and you'll get most paying customers to pay. Thieves don't generate revenue but paying customers do, so do your best to make it easy for them to pay. That's my two cents, anyway. I'm definitely interested in other viewpoints, contrary or otherwise. This discussion is definitely an important one for OSS. -Michael ___ -- 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] Realtime status feature - user feedback needed
On Wednesday 07 May 2008 16:11:05 Atis Lezdins wrote: However I encountered a resistance from Asterisk developers, as they don't wish to accept my patches - because they don't wish to support another interface. As I read in Bug Guidelines, if enough users request this, it should make into asterisk, so I'm asking You to express Your opinion on those features. That's not quite correct, either. First of all, the correct forum for this is the -dev list, where we discuss development issues. Second, we gave you an alternative way to do this. You could do this with AMI, with the addition of a single query to access current state, then monitor status continuously for updates. And third, it doesn't make a difference how many users request a particular interface -- the development team has to maintain it afterwards, and if you're proposing a new interface, you need to convince the development team that it's worth the extra hassle -- not the users. So we're not opposed to the concept; we are opposed to the particular interface that you chose to use. Modify it, and it will make its way back into Asterisk. Stubbornly stamping your foot and insisting that you have the right way, and the status quo will remain. *** Supporting this feature If You find that those features would be good for merging into Asterisk, please write a comment in bugtracker: http://bugs.digium.com/view.php?id=12556 Please don't. We've already discussed this to enough detail, and if you choose to modify your code, it will show up in the next major release of Asterisk. -- Tilghman ___ -- 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] Cisco to Asterisk migration
Thanks Adrian and all the other guys that gave me tips on this Femi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Marsh Sent: 07 May 2008 17:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco to Asterisk migration Basic process: 1) Build the A*k server so that it has tftp installed (or another box that does) 2) Build up the SIPdefault.conf and get the firmware files in place (see Cisco docs on this, plus theres loads on the wikis). 3) Test with a single phone, change its tftp server to the asterisk. Check that : a) The firmware switches to SIP b) the phone registers to A*k and all is well. Calls can be made etc... 4) Once your happy with the A*k config and I mean ***really*** happy, then add in all the configs for the other phones (I used scripts to build mine). 5) Try a few more phones manually. But eventually just update DHCP so that the tftp server option points to the A*k server. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Femi Sent: 25 April 2008 10:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Cisco to Asterisk migration Hi Guys, I have client with a Cisco 2690 call manager solution that wants to upgrade but cannot stomach the costs of continuing with Cisco The installation will go up to 100 users The client currently has about 40 Cisco phones and would like to continue with these phones with the odd Polycom I'm looking at plugging in an Asterisk box and using the existing Cisco box as a PSTN gateway only Has anyone on the list done this? Any pitfalls or tips you would like to share? Thanks Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] show CODEC in CDR
Antoine Megalla schrieb:
In asterisk is there a way of saving the voice codec used in the call in the
CDR.
I am having mostly SIP calls, with few H323 calls.
I have been trying for the past 2 weeks to figure it out on my own, but with
no luck.
There are no channel variables that can give the current codec used in the
call, or used in the channel.
I need it because I must charge the clients according to the codec for the
individual calls.
I even tried looking in the AGI commands for asterisk, and many external AGI
packages, but with no luck.
What happens if you put something like this in a macro
Verbose(1,${CHANNEL(audioreadformat)});
Verbose(1,${CHANNEL(audiowriteformat)});
Verbose(1,${CHANNEL(audionativeformat)});
and call the macro from Dial()
Dial(...,,M(my-codec-log-macro));
Grüße,
Philipp Kempgen
--
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amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Realtime status feature - user feedback needed
On Thu, May 8, 2008 at 1:07 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 07 May 2008 16:11:05 Atis Lezdins wrote: However I encountered a resistance from Asterisk developers, as they don't wish to accept my patches - because they don't wish to support another interface. As I read in Bug Guidelines, if enough users request this, it should make into asterisk, so I'm asking You to express Your opinion on those features. That's not quite correct, either. First of all, the correct forum for this is the -dev list, where we discuss development issues. Second, we gave you an alternative way to do this. You could do this with AMI, with the addition of a single query to access current state, then monitor status continuously for updates. And third, it doesn't make a difference how many users request a particular interface -- the development team has to maintain it afterwards, and if you're proposing a new interface, you need to convince the development team that it's worth the extra hassle -- not the users. True, but resistance I encountered gave me impression that there's no way how to convince devs except lot of users asking for this, so i want to see who would find this useful. I hope that this would convince the development team. So we're not opposed to the concept; we are opposed to the particular interface that you chose to use. Modify it, and it will make its way back into Asterisk. Stubbornly stamping your foot and insisting that you have the right way, and the status quo will remain. Unfortunately the concept I'm offering is that There's no need for continuous AMI connection. Current state can be retrieved already (but that needs locking), and incremental updates are available too (but that needs continuous AMI connection). So all together - I'm saying there could be really simple interface for all this - no troubles with locking of lists or keeping persistent connections. Why would user application need to take care of all this, if DB engine can do that. *** Supporting this feature If You find that those features would be good for merging into Asterisk, please write a comment in bugtracker: http://bugs.digium.com/view.php?id=12556 Please don't. We've already discussed this to enough detail, and if you choose to modify your code, it will show up in the next major release of Asterisk. I understand that code have to match certain standards, coding guidelines and architecture. I'm willing to do any of this, but so far i see all the discussions are about concept. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] URGENT
Similar system and situation here I finally taught mine to log off and on without logging in phantom extensions or external numbers (THAT was almost a disaster) But prior to that Do Not Disturb helped except for not allowing internal, non-queue calls to be answered without going to VM. I'm still getting some grief about the Poycom's lack of programmable feature buttons (like log in / out) but it's working. Dale On Wed, May 7, 2008 at 4:49 PM, Tarek Sawah [EMAIL PROTECTED] wrote: Hello, I have given up hope of finding a solution for my problem and I think this is my last resort. At my company I have a Trixbox box and I used freePBX to configure the pbx. They have a queue with 15 static members.. they are not online all the time.. still when ever a call comes in the queue rings ALL members without skipping unavailable ones offline ones which creats so much load on the system. I can't teach the agents to REMEMBER logging off the queue and logging in .. so Static agents is what I have to do.. now it is a pain in the butt to manually login and logout agents and I can't get the agents.conf thing work properly or work at ALL.. Is there anyway that the queue senses the offline users and logthem off or at least skip them without the need for a manual operation? Tarek Sawah Technical Advisor ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX issues with 1.4.19.1
Brian J. Murrell wrote: Right. Which to me at least, tightly couples it. IOW, the security fix, while yes, it fixes the security problem, is quite useless without this other fix as it makes iax2 unstable. I agree with you. I am in the process of working on the Asterisk 1.2.20 release, which will contain this IAX2 fix. However, you have convinced me that this fix should be released against the previous security release, as well. So, tomorrow, I will make an Asterisk 1.4.19.2 release, which includes this one change. As a part of making that release, it will come with a patch against 1.4.19.1, which will include the changes needed to make IAX2 usable again, if people needed the patch for a custom version. I will also update the security advisory to note the effects of the original changes to address the security issue. I will then publish announcements as usual that to hopefully notify everyone that doesn't closely monitor commits to Asterisk, or other high volume Asterisk mailing lists. Thanks for the feedback, -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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
[asterisk-users] Big difference in CPU utilization with MeetMe
Hello everyone, We are building a new * server based on a Supermicro motherboard with a 2.8 Xeon processor and a TE220B card. We're using the PBX In a Flash distribution. What we've found is that with a 4 user MeetMe conference, the CPU usage is consistently around 16%. This in comparison to our existing PSTN gateway * box running 1.09 (it hosts our conferences and terminates our T1s). With 23 users and processing all PSTN phone calls, CPU usage averaged from 3-8%. This is an older Supermicro, with a 2.4 Xeon processor. In both cases, the connections are via IAX trunks from our main PBX here, and in two remote locations. We use g711 u-law only - no other codecs are used. If we connect the same number of users through a PRI connection directly to the new server, the CPU is 1% or less, so obviously we've pooched something. We saw this same behavior when we split off the users to a 1.4x based PBX, and we thought it was the server hardware in the new machine, which was an older Dell 2650. But now we're not so sure. I know this is kind of vague, but can anyone suggest what might be happening? New Server CentOS 5, Kernel version 2.6.18-53.1.14.el5 Asterisk 1.4.19.1, and the SVN Zaptel drivers for the TE220B problems posted recently 2.8 Xeon, Hyperthreading disabled, 4GB RAM, 3Ware 9550SX RAID Old Server Fedora, Kernel version 2.4.22-1.2199.nptl Asterisk 1.0.9 2.4 Xeon, Hyperthreading off, 1GB RAM Thanks for the help, Kevin ___ -- 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] DUNDi call impossible in one direction
Andrea Spadaccini wrote: I've set up DUNDi between two asterisk boxes, and sometimes happens that calls from machine A can't reach peers in machine B, but calls from B to A work correctly. The strange thing is that the CLI command 'dundi show peers' shows correctly the registered peer in both servers, and in this situation if I make a call from B to A, suddenly peers in server A are able to call peers in machine B. Try using the DUNDi query CLI command to see what results your server is getting when you try to make calls. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] Basic modules of Asterisk
Sanjay Rajdev wrote: I just want to Run Asterisk with the basic required modules, What can I do to achieve so? My only requirement is to run SIP clients and the Dictate Module. 2 options: 1) Before compiling and installing Asterisk, run make menuselect to select only the modules that you want to use. That way, only those modules are compiled and installed. 2) After installing Asterisk, edit /etc/asterisk/modules.conf. By default, Asterisk will load all installed modules. You can turn off the autoload functionality, and explicitly list the modules that you need. You probably want pbx_config, chan_sip, app_dictate, app_dial, probably some others ... -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] T38 Passthrough Verification
JR Richardson wrote: I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet ! sip show channels shows the call setup with ulaw. Try setting canreinvite=no for the peer doing T.38. It looks like the code in Asterisk 1.4 will not allow re-invites for an established T.38 passthrough call. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] Big difference in CPU utilization with MeetMe
There is a bug in 1.4.19.1 with IAX. That's your issue. On Wed, May 7, 2008 at 12:38 PM, Kevin Ragsdale [EMAIL PROTECTED] wrote: Hello everyone, We are building a new * server based on a Supermicro motherboard with a 2.8 Xeon processor and a TE220B card. We're using the PBX In a Flash distribution. What we've found is that with a 4 user MeetMe conference, the CPU usage is consistently around 16%. This in comparison to our existing PSTN gateway * box running 1.09 (it hosts our conferences and terminates our T1s). With 23 users and processing all PSTN phone calls, CPU usage averaged from 3-8%. This is an older Supermicro, with a 2.4 Xeon processor. In both cases, the connections are via IAX trunks from our main PBX here, and in two remote locations. We use g711 u-law only - no other codecs are used. If we connect the same number of users through a PRI connection directly to the new server, the CPU is 1% or less, so obviously we've pooched something. We saw this same behavior when we split off the users to a 1.4x based PBX, and we thought it was the server hardware in the new machine, which was an older Dell 2650. But now we're not so sure. I know this is kind of vague, but can anyone suggest what might be happening? New Server CentOS 5, Kernel version 2.6.18-53.1.14.el5 Asterisk 1.4.19.1, and the SVN Zaptel drivers for the TE220B problems posted recently 2.8 Xeon, Hyperthreading disabled, 4GB RAM, 3Ware 9550SX RAID Old Server Fedora, Kernel version 2.4.22-1.2199.nptl Asterisk 1.0.9 2.4 Xeon, Hyperthreading off, 1GB RAM Thanks for the help, Kevin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mediatrix 2102's
It was a big pain for me, I change all to Linksys spa2102 I had 20 Mediatrik as a paper weight Sorry Polo -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Daniel Lynes Enviado el: Miércoles, 07 de Mayo de 2008 04:37 p.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Mediatrix 2102's Hello all. I'm encountering an issue whereby a Mediatrix 2102 is able to register, authenticate, and place a call into an asterisk box. However, the problem happens when the asterisk box tries to proxy the call to another Mediatrix 2102, or back to the other port on the same Mediatrix 2102. No call progression, and times out trying to send a message back to the Mediatrix, or to a new one. Please find attached a wireshark packet capture of the issue. If anyone could give me an idea as to where I can start looking to solve this issue, or if they have some experience in getting Mediatrix devices talking to each other through Asterisk, it would be greatly appreciated. Thanks. ___ -- 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] Realtime status feature - user feedback needed
On Wednesday 07 May 2008 17:27:33 Atis Lezdins wrote: So all together - I'm saying there could be really simple interface for all this - no troubles with locking of lists or keeping persistent connections. Why would user application need to take care of all this, if DB engine can do that. Your question leads to this question: why don't you create a proxy application that listens on AMI and populates a database outside of Asterisk, then do all your queries to that database? That would provide exactly the same functionality, but it would not require a single change to the Asterisk codebase. You could even contribute that application back as something in the contrib/scripts subdirectory. -- Tilghman ___ -- 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] update DB on ringing/ catch ringing event
On Thu, May 08, 2008 at 12:19:52AM +0300, Atis Lezdins wrote: On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: Benjamin Jacob schrieb: Anyway in Asterisk to update a DB/ do some action on events like ringing. The issue is I need to be able to hangup/cancel a call, if it's ringing(decided by the admin). This is independant of the timeout that we can specify in the Dial command. If I could somehow update a DB with the channel name on ringing, it would solve my problem. I assume NVlinedetect is one way to do it, but that isn't visible anymore, more so for Asterisk 1.4 and above. Any bright ideas on this one? I think there is no other solution but to listen to events on the Asterisk manager interface. For now, not really. You could try Realtime Channels patch I just mentioned here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211136.html This should give you up-to-date list of channels in database, so you can use SELECT * FROM channels WHERE state=Ring; to get currently ringing channels. If You find this patch useful, please add a comment to issue http://bugs.digium.com/view.php?id=12556 that you would like to see Realtime status implemented in future versions of Asterisk. So you constantly poll the status of all channels? Waiting on manager interface event sounds more effective to me. But what exact ringing is it? Isn't the call by then already in the dialplan (and could be hung up before answered?) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime status feature - user feedback needed
Tilghman Lesher a écrit : On Wednesday 07 May 2008 17:27:33 Atis Lezdins wrote: So all together - I'm saying there could be really simple interface for all this - no troubles with locking of lists or keeping persistent connections. Why would user application need to take care of all this, if DB engine can do that. Your question leads to this question: why don't you create a proxy application that listens on AMI and populates a database outside of Asterisk, then do all your queries to that database? That would provide exactly the same functionality, but it would not require a single change to the Asterisk codebase. You could even contribute that application back as something in the contrib/scripts subdirectory. I second that, If there is already a way to do things, why adding another one, especialy if it's for caching reasons. While we cannot say that asterisk fall into the KISS rule, it's not a reason to let it grow. -- Benoit Plessis +33 6 77 42 78 32 [EMAIL PROTECTED] +33 4 67 28 06 96 ___ -- 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] Out-Going Calleriid
Tim Guy wrote:
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten = _9.,1,Set(CALLERID(number)=01926xx)
exten = _9.,2,Dial(ZAP/1/${EXTEN:1})
Ive tried all permutations of the CALLERID (ie CALLERID(NAME) and
CALLERID(NUMBER) but it just wont work anymore.
Zapata has the following relevant settings
usecallerid=yes
hidecallerid=no
callwaiting=yes
you may need to add
usecallingpres=yes
in zapata.conf
and also add
exten = _9.,n,SetCallerPres(allow)
before the Dial command in extensions.conf.
--
Edwin Lam [EMAIL PROTECTED]
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] Realtime status feature - user feedback needed
On Thu, May 8, 2008 at 1:23 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Tilghman Lesher a écrit : Your question leads to this question: why don't you create a proxy application that listens on AMI and populates a database outside of Asterisk, then do all your queries to that database? That would provide exactly the same functionality, but it would not require a single change to the Asterisk codebase. You could even contribute that application back as something in the contrib/scripts subdirectory. I second that, If there is already a way to do things, why adding another one, especialy if it's for caching reasons. While we cannot say that asterisk fall into the KISS rule, it's not a reason to let it grow. Agreed. There should be ONE to do it, it should be SIMPLE and as RELIABLE as possible, without interfereing (bad spelling?) with asterisk's operations: the proxy into AMI looks like the way to acheive the required funcionality... After all, that's exactly the purpose of AMI ! Let's keep the codebase as small as possible, let's make asterisk as solid and reliable as possible. Let's not reinvent wheels! -- exvito ___ -- 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] Realtime status feature - user feedback needed
On Thu, May 8, 2008 at 3:49 AM, Ex Vito [EMAIL PROTECTED] wrote: On Thu, May 8, 2008 at 1:23 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Tilghman Lesher a écrit : Your question leads to this question: why don't you create a proxy application that listens on AMI and populates a database outside of Asterisk, then do all your queries to that database? That would provide exactly the same functionality, but it would not require a single change to the Asterisk codebase. You could even contribute that application back as something in the contrib/scripts subdirectory. True, that was one of initial options, however I prefer to NOT have yet another layer. I will consider this as an option where appropriate. However this looks quite awkward to me, somehow it reminds me tailing queue_log or CDR and putting result into MySQL database.. just one level more that way. For now, I see only one point against this - having status cleared upon module load/unload makes it easier to follow restarts/module loads. I second that, If there is already a way to do things, why adding another one, especialy if it's for caching reasons. While we cannot say that asterisk fall into the KISS rule, it's not a reason to let it grow. Agreed. There should be ONE to do it, it should be SIMPLE and as RELIABLE as possible, without interfereing (bad spelling?) with asterisk's operations: the proxy into AMI looks like the way to acheive the required funcionality... After all, that's exactly the purpose of AMI ! Let's keep the codebase as small as possible, let's make asterisk as solid and reliable as possible. Let's not reinvent wheels! Ok, so we're exactly at the point. Yes, I agree that it would act nearly the same way as AMI actions, however there's one great advantage - It would be really easy to set this up for user. AMI proxy would take more effort, need configuration, etc. Then there should be much more development support for proxy than for code within asterisk (if you have noticed, there's no new code, just reusing existing functionality) I think that there should be several ways how to do something, not just one. Having realtime status won't mean that much changes, for now I can see only 4 families for this - queue_members (already existing), queue_callers, channels and meetme. Really nothing more to give full overview of Asterisk Status. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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
[asterisk-users] dundi network - redundancy / fault tolerance ?
Hi list, I'm planning a private DUNDi network for a cross-country distributed PBX. Initially it will be composed of about 10 systems, growing to about 20. Current requirements point to a topology of two interconnected DUNDi hubs, each peering with half the PBXs... This would lead to two interconnected / inter-peered stars. Example: - Consider PBXs A to H - C and E will be hubs and peer with each other - A, B and D peer with C - F, G and H peer with E This leads to a maximum three hop lookup and will make good use of current network topology / bandwidths. Of course, should any of the hubs be unavailable and the lookup capability is severely compromised. Now, how to move on to acheive some kind of fault tolerance ? According to the docs we've studied, DUNDi does not like loops (which we assume one can limit with low enough TTLs). Our doubts are: - Should one use the order peer parameter to specify alternate lookup paths / peers ? Is that its purpose ? If not, what is it used for ? - Alternatively, should one create loops in the DUNDi topology and limit them via TTL ? - If both options are possible, which would be the trade-offs between them ? (Not clear at all to us!) - Assuming any of the above is possible as a means to acheive redundancy, which of the following topologies would your prefer ? (hmmm, maybe I need to refresh my graph theory...) ;-) #1 - Peer each PBX with both hubs #2 - Duplicate both hubs and peer each PBX with its hub and its hub dup For better understanding, take a look at: #1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png #2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png Thanks in advance for review and feedback. Cheers, -- exvito ___ -- 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] Newbie alert: VoIP hardware
On Wed, May 7, 2008 at 11:38 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 07 May 2008 09:40:21 Steve Totaro wrote: Interesting results in Google for TDM400P TigerJet reference design. http://www.google.com/search?hl=ensafe=offclient=firefox-arls=org.mozill a:en-US:officialhs=h9Ppwst=1sa=Xoi=spellresnum=1ct=resultcd=1q=Tiger jet+Reference+design+tdm400pspell=1 Other keywords turn up much more similar results that seem to confirm that the reference design from TigerJet was used. As with anything on the internet, take it with a grain of salt but it does have enough hits to raise questions. No, it doesn't. It simply is an oft-repeated falsehood. GO to the TigerJet page, LOOK at the reference designs. They do not hide a single reference design from the web, and NONE of them are the TDM400P design. If it was a reference design, please show the world the reference design from TigerJet. There simply isn't one, and repeating it does not make it so. -- Tilghman I guess you have reading comprehension issues. I said take it with a grain of salt as well as seems to confirm. Both are very benign and offer two sides of the story. I think your personal feelings are overpowering your ability to comprehend and reason. Anyways, maybe the entire reference design is not there but just connect the 2-3 reference designs and you're there. Tigerjet provides the reference design of using the PCI chipset + they provide the reference design of the X100P (pretty much) and going from X100P to PCI card with one FXS module is not that hard (just different Silabs chip) and then multiplying it only needs a small CPLD chip. Not much brain power to come up with that. BTW, rumor has it that Mark Spencer did not have contracts for employees with the exception of salary, back in the old days. Maybe you can ask him or check Martin's employee file. Thanks, Steve Totaro ___ -- 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] dundi network - redundancy / fault tolerance ?
Ex Vito wrote: Now, how to move on to acheive some kind of fault tolerance ? According to the docs we've studied, DUNDi does not like loops (which we assume one can limit with low enough TTLs). Which documentation are you referring to? You may have misunderstood something, or there may be some false information floating around the internet (*GASP*). The DUNDi protocol has built in handling for loops. It keeps track of which nodes have already been queried, so you don't have to worry about loops in your network. Every node can peer with every other node if you really wanted to. Of course, that's not necessarily the most efficient thing to do ... Our doubts are: - Should one use the order peer parameter to specify alternate lookup paths / peers ? Is that its purpose ? If not, what is it used for ? The order parameter is really a tool. There is not an exact situation that it is intended for. It depends on your network. Keep in mind that DUNDi caches results along the way. If you use the order option to have servers send queries through a primary server, you getter better caching performance. - Alternatively, should one create loops in the DUNDi topology and limit them via TTL ? As I said before, don't worry about loops. Set your TTL to handle a worst case path for a query in your DUNDi topology. - If both options are possible, which would be the trade-offs between them ? (Not clear at all to us!) I'm not sure what you mean. The best thing to do is to have multiple peers. Have every server have at least two peers. Setting a primary and secondary can be good for caching reasons. - Assuming any of the above is possible as a means to acheive redundancy, which of the following topologies would your prefer ? (hmmm, maybe I need to refresh my graph theory...) ;-) #1 - Peer each PBX with both hubs #2 - Duplicate both hubs and peer each PBX with its hub and its hub dup For better understanding, take a look at: #1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png #2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png Thanks in advance for review and feedback. I'm not necessarily up on my graph theory, either, but I would probably go with something like #1. A combination of having multiple peers and usage of the order option can give you good redundancy without hurting your performance. When you set primary, secondary, etc. peers, the server will attempt to contact them one at a time. If you have multiple peers, but do not set an order, they will all be contacted at once, which may (probably will) increase latency for call completion, will increase bandwidth consumption, among other things. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] dundi network - redundancy / fault tolerance ?
I don;t have any answers for you... But I would love to hear about the results after you get this working and what road blocks you hit and how you overcame them. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Ex Vito [EMAIL PROTECTED] Sent: Wednesday, May 07, 2008 10:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] dundi network - redundancy / fault tolerance ? Hi list, I'm planning a private DUNDi network for a cross-country distributed PBX. Initially it will be composed of about 10 systems, growing to about 20. Current requirements point to a topology of two interconnected DUNDi hubs, each peering with half the PBXs... This would lead to two interconnected / inter-peered stars. Example: - Consider PBXs A to H - C and E will be hubs and peer with each other - A, B and D peer with C - F, G and H peer with E This leads to a maximum three hop lookup and will make good use of current network topology / bandwidths. Of course, should any of the hubs be unavailable and the lookup capability is severely compromised. Now, how to move on to acheive some kind of fault tolerance ? According to the docs we've studied, DUNDi does not like loops (which we assume one can limit with low enough TTLs). Our doubts are: - Should one use the order peer parameter to specify alternate lookup paths / peers ? Is that its purpose ? If not, what is it used for ? - Alternatively, should one create loops in the DUNDi topology and limit them via TTL ? - If both options are possible, which would be the trade-offs between them ? (Not clear at all to us!) - Assuming any of the above is possible as a means to acheive redundancy, which of the following topologies would your prefer ? (hmmm, maybe I need to refresh my graph theory...) ;-) #1 - Peer each PBX with both hubs #2 - Duplicate both hubs and peer each PBX with its hub and its hub dup For better understanding, take a look at: #1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png #2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png Thanks in advance for review and feedback. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Restarting due to segfault
Sanjay Rajdev wrote: I have Asterisk 1.4.15 installed on a Fedora Core 8 machine. Asterisk is snip In the dialplan we have used MixMonitor() to record the calls. Can anyone help me on getting to the root of the problem or fixing it? We have fixed a _lot_ of issues in that area of the code since 1.4.15. I would suggest trying the latest version. If it still gives you trouble, please let us know on http://bugs.digium.com so that we can fix it up for you. Thanks, -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] DUNDi call impossible in one direction
Are you using IAX2 as your transport between the 2 servers or SIP? If you are using IAX2, are you using Asterisk 1.4.18.1 or 1.4.19.1 on either machine? If so, you may be encountering the IAX2 bug that some have been discussing on the list recently you can read it here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211000.html -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Russell Bryant [EMAIL PROTECTED] Sent: Wednesday, May 07, 2008 6:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi call impossible in one direction Andrea Spadaccini wrote: I've set up DUNDi between two asterisk boxes, and sometimes happens that calls from machine A can't reach peers in machine B, but calls from B to A work correctly. The strange thing is that the CLI command 'dundi show peers' shows correctly the registered peer in both servers, and in this situation if I make a call from B to A, suddenly peers in server A are able to call peers in machine B. Try using the DUNDi query CLI command to see what results your server is getting when you try to make calls. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie IVR: How to read() before playback() is finished?
Besides the Background() app mentioned, you might like the WaitExten()
app
Thanks guys for your response.
I have had much success with Read() as below so that whenever I press a
key before the sound file finishes playing, it will read the digit and
move to the next line.
exten = 100,1,Answer()
exten = 100,n,Read(OPTION,SOUND-FILE,1)
exten = 100,n,GotoIf($[${OPTION} = 2]?do2:doothers)
[...]
However, I noticed that sometimes when I call from the outside line to
this number, I need to press the key many many times before the digit
can be read. This does not happen if I do it on the LAN.
Is there any way I could fix this problem?
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