RE: [asterisk-users] play music while continue executing dial plan
You are better off running a small AGI script and calling the Dialplan functions from there. You can always start musiconhold, process, and return to dial plan. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rilawich Ango Sent: Monday, January 15, 2007 2:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] play music while continue executing dial plan It doesn't work as it will hold up the call without running the rest of the statement. On 1/12/07, Octavio Ruiz (Ta^3) [EMAIL PROTECTED] wrote: Perhaps MusicOnHold() app? Is there any application can let the dial plan to execute while playing music? Say I have a lot of command to do in the dial plan but I don't want to keep silence while execution of dial plan. I notice Background(file) can play music but it will hold until pressing a key. I want something like background and it plays music with continuing execute the rest of the command in dial plan. -- YOW!!! I am having fun!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE : TDM2400p bad sound quality
Uhm. Actually if I write: cat /proc/interrupts I get: 11: 2997154835 XT-PIC libata, wctdm24xxp Is this the problem? How can I solve it? The output of zztest is: [EMAIL PROTECTED] freepbx-2.2.0]# zttest Opened pseudo zap interface, measuring accuracy... 100.00% 100.00% 99.987793% 100.00% 100.00% --- Results after 5 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.997559 Thank you very much indeed! Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE : TDM2400p bad sound quality
On Mon, 15 Jan 2007, Giuffredi wrote: Uhm. Actually if I write: cat /proc/interrupts I get: 11: 2997154835 XT-PIC libata, wctdm24xxp Is this the problem? Potentially yes. The 2400 card is sharing interrupts with the IDE disk system. How can I solve it? Try moving the card to another slot in the PC. See if the PCs BIOS has options to reserve, or fix IRQs to a particular slot. The output of zztest is: --- Results after 5 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.997559 Which looks good - and you'll probably be fine - until you do lots of disk IO at the same time, but if you can move the card to a different IRQ, that would be a good starting point. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime and MD5 authentication
Hi, I've troubles with setting up Asterisk Realtime and MD5 authentication. With clear text passwords everything is working fine. -- Registered SIP 'edwin' at 10.0.0.37 port 5060 expires 600 -- Saved useragent Cisco-CP7940G/8.0 for peer edwin [2007-01-15 10:18:12] DEBUG[28528]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [2007-01-15 10:18:12] DEBUG[28528]: res_config_mysql.c:355 update_mysql: MySQL RealTime: Update SQL: UPDATE SipUser SET ipaddr = '10.0.0.37', port = '5060', regseconds = '1168853292', username = 'edwin', fullcontact = 'sip:[EMAIL PROTECTED]:5060;transport=udp' WHERE name = 'edwin' [2007-01-15 10:18:12] DEBUG[28528]: res_config_mysql.c:369 update_mysql: MySQL RealTime: Updated 1 rows on table: SipUser [2007-01-15 10:18:13] NOTICE[28528]: chan_sip.c:12001 handle_response_peerpoke: Peer 'edwin' is now Reachable. (328ms / 2000ms) After changing clear text passwords in MD5, and rename the md5-field in the database to 'md5secret' no SIP account can register. [2007-01-15 11:06:23] DEBUG[28528]: db.c:236 ast_db_del: Unable to find key 'edwin' in family 'SIP/Registry' [2007-01-15 11:06:23] DEBUG[28521]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [2007-01-15 11:06:23] DEBUG[28521]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM SipUser WHERE name = 'edwin' [2007-01-15 11:06:23] DEBUG[28521]: acl.c:199 ast_append_ha: 0.0.0.0/255.255.255.255 appended to acl for peer [2007-01-15 11:06:23] DEBUG[28521]: acl.c:199 ast_append_ha: 10.0.1.77/255.255.0.0/255.255.0.0 appended to acl for peer [2007-01-15 11:06:23] DEBUG[28521]: db.c:197 ast_db_get: Unable to find key 'edwin' in family 'SIP/Registry' I've added user:[EMAIL PROTECTED] in the auth-field and created the md5-password (in the field md5secret) as follow: md5 -s edwin:[EMAIL PROTECTED] (with secret and realm as the correct values). I'm using Asterisk 1.4.0. -- Edwin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To 1.4 or not
I agree with C F - We just upgraded to our first non-internal 1.2.x system last Friday. Mostly I am glad we waited. I imagine we may upgrade to 1.4 in about a year :) Really it depends on your customer. If it is a commercial operation I would be cautious of 1.4 still, and at the very least test it thoroughly with the hardware and configuration you plan to use. On the other hand in a less critical environment, exercising 1.4 is a good thing for all - We all know there are bugs in there, and you are helping to find them... Cheers, S. On 1/15/07, C F [EMAIL PROTECTED] wrote: Change log can help you a lot. I would stick to my grandmothers advice, if it aint broken don't fix it. On 1/14/07, Yuan LIU [EMAIL PROTECTED] wrote: I don't have a particular reason to upgrade, but I'm installing a new box, so I have the opportunity to go 1.4. On the other hand, I'm not familiar with 1.4, and relatively new to Asterisk. So instead of trying to keep up with two different versions, I want to tie my handful of boxes to one, before any of them grow too complex. Is there a document about the main motivations to upgrade? From your practice, what are your primary reasons? Thank you in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] phpagi transfer example
Hi, i want to to this thing with php AGI:
#!/usr/local/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
$agi-answer();
$cid = $agi-parse_callerid();
$agi-text2wav(Hello, {$cid['name']}.);
$agi-text2wav('Enter some numbers and then press the pound key. Press
1 1 1 followed by the pound key to quit.');
$result = $agi-get_data('beep', 3000, 20);
$keys = $result['result'];
$agi-text2wav('You will b transfered to $keys');
//transfer to $keys
?
Ok, how can i do the transfer from the caller to $keys ?
Thanks
___
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RE: [asterisk-users] OT: Quad-band cellphones with wifi stablesipsupport
Its not quad band and in my opinion doesn't perform well enough to be used for anything but basic email and phone calls. This phone, even on the newest version of firmware (Sprint) hangs when syncing with exchange to the point where you miss calls even though you tried to answer them. If you turn on wifi or Bluetooth, it simply compounds the problems. It will also require (literally) a ritualistic daily reset Notice the reset button on the bottom of the phone? Seriously, unless you live in an area where EVDO isn't offered even at 1x, forget the wifi. I don't think this phone has the muscle you want. My Treo 700wx outperforms my old PPC-6700 3 to 1 and doesn't lockup or need reset (rarely). The only really feature I lost was wifi and it took me two weeks to even notice this phone didn't have it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Norton - SophMedia LLC Sent: Monday, January 15, 2007 12:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Quad-band cellphones with wifi stablesipsupport Hey Tomer, I'm not sure if the Audiovox PPC6700 is quad band, but it does support Wifi and runnings SJPhone great! It is even usable over Sprints EVDO service. On Mon, 15 Jan 2007 08:01:44 +0200, Tomer Horn [EMAIL PROTECTED] wrote: Hello, I am looking to purchase a new quad-band cellphone and I'm looking for one with WiFi and enough CPU power for stable SIP calls. I was wondering if anyone here can share his experience and recommend on a good cellphone. Any chance there is such a phone with even good WiFi profiles management or am I asking for too much now? :-) Thanks, Tomer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rt db lookup
Which command effects whether or not the * server will lookup a peer from the db even though the phone isn't registered locally? I have several * servers but I want any server to be able to lookup and send a call to phones registered on another server (SIP cluster?). Thanks Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 and sip list peers
Hi, I have not checked this, but I thought the intention was that 'show' was a human readable formatted output, and 'list' was meant to be the same data but more easily machine readable. Of course I could be completely wrong. Steve On 1/13/07, Jerry Geis [EMAIL PROTECTED] wrote: I thought I read where 1.4 changed sip show peers to sip list peers. the help is still showing sip show peers. Did it change back? Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing Asterisk 1.4 Documentation
Is there an official list anywhere specifying the Prerequisites for installing asterisk(Specifally 1.4) on Fedora Core 4? I have been struggling with a configure: error: termcap support not found error when compiling 1.4 on my brand new install of FC4 fully updated, Fedora was installed as a base install to try and keep the overheads down of running a full GUI and all the other junk, but was lacking the ncurses-devel package. Once all of the prereq's were installed it compiled fine, its frustrating I cant find a This is what you must have installed before beginning your Asterisk install' Thanks Cory ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possibility to catch DTMF when 2 users are in a conversation
I will sum up the results of my investigations : - When canreinvite is set to yes, I manage to make a video call between the 2 parties, when I emit a DTMF signal, it triggers the playback of a sound clip correctly, but I can't playback a video clip. - When canreinvite is set to no, The DTMF I emit is not detected by Asterisk, although I see the SIP INFO message in the SIP debug messages of Asterisk. I copy in line the relevant abstracts of my configuration files : -- sip.conf -- [8160] type=friend username=8160 secret=** host=dynamic context=default disallow=all allow=ulaw allow=h263 dtmfmode=info canreinvite=yes insecure=very -- features.conf -- [applicationmap] test = 9,peer,Playback,hello-world ; TEST with sound clip testVideo = 8,peer,Playback,/tmp/test ; TEST with video clip -- extensions.conf -- exten = 8160,1,Set(DYNAMIC_FEATURES=test#testVideo) exten = 8160,n,Dial(SIP/8160) 2007/1/14, Andrew Joakimsen [EMAIL PROTECTED]: What video clip? Does a native video call between the two work? On 1/14/07, Antoine Fressancourt [EMAIL PROTECTED] wrote: Le 13 janv. 07 à 02:10, Leo Ann Boon a écrit : Antoine Fressancourt wrote: Hello, Thank you Leo for your answer, I manage to do what I want perfectly when both the caller and the callee are set in SIP with canreinvite=no using SIP INFO method for DTMF. Now, I can't figure out why this can't work when I set canreinvite = yes with the same DTMF method. Running Wireshark on my machine, I see that the SIP INFO messages are sent to the Asterisk box running as a proxy, but the INFO message doesn't trigger any action. Relooking at your requirements, I'd say you must use canreinvite=no. Otherwise, there's no way for Asterisk to inject audio into the stream. I tried to set canreinvite=no, but the problem is that Asterisk can't do the reinvitation to the callee in order to send the video clip. Is there a way to allow asterisk to do such a reinvitation in order to make it play the video stream correctly ? Thank you very much for your help. Antoine ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wanpipe 2.3.4-2 + kernel 2.6.19 = problems
Hi list. some info: zaptel 1.4.0 wanpipe 2.3.4-2 kernel 2.6.19.1 Debian I'm trying to build wanpipe on my server, but I got a error that it can't find config.h.. I found a post on an other unrelated mailing list which stated that includes/linux/config.h has been removed from 2.6.19. It also suggested replacing all references to config.h with autoconf.h make -C /lib/modules/2.6.19.1/build SUBDIRS=/usr/src/wanpipe/kdrvtmp CC=gcc KBUILD_VERBOSE=0 modules make[1]: Entering directory `/usr/src/linux-2.6.19.1' CC [M] /usr/src/wanpipe/kdrvtmp/sdladrv_src.o In file included from /usr/src/wanpipe/kdrvtmp/sdladrv_src.c:131: include/linux/wanpipe_includes.h:226:63: linux/config.h: No such file or directory make[2]: *** [/usr/src/wanpipe/kdrvtmp/sdladrv_src.o] Error 1 make[1]: *** [_module_/usr/src/wanpipe/kdrvtmp] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.19.1' make: *** [all] Error 2 This is the error i got. I've grepped through all of my include/linux/ wanpipe_includes.h files i have on my server (there is actually a couple of them), and replaced config.h with autoconf.h, but still i get the same error. Looks like I'm unable to locate the include/linux/ wanpipe_includes.h file wanpipe is actually looking for. Is there a patch or a newer version of wanpipe that has this issue solved? Erik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re:Nat Question
Thanks for help me, well, I do all that i see on the wiki page about asterisk and nat troubleshooting, because did not work I connected asterisk to a public ip for testing, but, while I get two sip phones with private ip connected to my asterisk with public ip, I can setup calls(phones rings) but I can't hear nothing from both phones. In this case which would be the problem?. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To 1.4 or not
Hello Yuan, I have recentky spoken to a number of customers who run call-centers, tried 1.4 test installs and concluded it's not there yet in terms of reliability. If I were to install a production box today, I would go for 1.2. l. In data Mon, 15 Jan 2007 00:01:27 +0100, Yuan LIU [EMAIL PROTECTED] ha scritto: I don't have a particular reason to upgrade, but I'm installing a new box, so I have the opportunity to go 1.4. On the other hand, I'm not familiar with 1.4, and relatively new to Asterisk. So instead of trying to keep up with two different versions, I want to tie my handful of boxes to one, before any of them grow too complex. Is there a document about the main motivations to upgrade? From your practice, what are your primary reasons? Thank you in advance. Yuan Liu -- Home of QueueMetrics - http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanpipe 2.3.4-2 + kernel 2.6.19 = problems
This is the error i got. I've grepped through all of my include/linux/ wanpipe_includes.h files i have on my server (there is actually a couple of them), and replaced config.h with autoconf.h, but still i get the same error. Looks like I'm unable to locate the include/linux/ wanpipe_includes.h file wanpipe is actually looking for. Is there a patch or a newer version of wanpipe that has this issue solved? From the changelog of 2.3.4-4 released on 2007-01-09 (ftp://ftp.sangoma.com/linux/current_wanpipe/ChangeLog.stable) - Updates for 2.6.18 and 2.6.19 kernels. hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phpagi transfer example
Ok, how can i do the transfer from the caller to $keys ? Probably by using a goto : http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI.html#goto hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MFC/R2 problems
These are the different meanings for the diferrent error codes: T1 TIMEOUT = 32769 T2 TIMEOUT = 32770 T3 TIMEOUT = 32771 UNEXPECTED MF SIGNAL= 32772 UNEXPECTED CAS = 32773 INVALID STATE = 32774 SET_CAS FAILURE = 32775 SEIZE ACK TIMEOUT = 32776 DEVICE IO ERROR = 32777 T1B TIMEOUT = 32778 I hope it helps. Greets On 1/8/07, yusuf [EMAIL PROTECTED] wrote: Hi, if that means I should post my config, here goes: zaptel: span=1,1,3,cas,hdb3,crc4 cas=1-15:1101 cas=17-31:1101 unicall.conf: protocolvariant=id,10,10 protocolend=cpe group=1 channel = 1-15 channel = 17-31 wanpipe1.conf FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CAS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO Josué Conti wrote: Hi Yusuf, how are you? It orders in the list its configurations, so that let us can help. Best Regards Josue 2007/1/8, yusuf [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Jan 8 13:04:11 DEBUG[12252]: chan_unicall.c:955 unicall_disable_ec: disabled echo cancellation on channel 1 Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] -- Hungup 'UniCall/1-1' What does - Unicall/1 protocol error. Cause 32769 mean, and can anyone help me. -- -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk 1.4 Documentation
Cory Hawkless wrote: Once all of the prereq’s were installed it compiled fine, its frustrating I cant find a “This is what you must have installed before beginning your Asterisk install’ It hasn't changed from Asterisk 1.2; termcap (ncurses or similar) is pretty much the only mandatory prerequisite; the others will just enable various parts of Asterisk to be built if they are found. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parked calls with Asterisk 1.4.0
Hi List. We have a small issue with making parked calls work with the new Asterisk 1.4. I have an impression that this used to work with 1.2, so its either I'm doing something wrong, or a regression. I hope its not the latter and you can tell me what I'm doing wrong. The setup is an Asterisk with sip users in mysql realtime and dialplan in mysql static (mostly - some stuff is built-in). We have Linksys hardware voip phones connected to it, and a small dundi setup (I don't think its important in this case). Here's the SIP users' default context: [local-priv-incoming] exten = 910,1,Goto(parkedcalls,700,1) parked calls looks like this, of course: CLI dialplan show parkedcalls [ Context 'parkedcalls' created by 'res_features' ] '700' = 1. Park() [res_features] So , supposedly someone calls me (in this case through the dundi setup, but I don't think its a problem - we can reproduce this with local calls as well) and I do attendant transfer to '900'. I then hear the parked call number (701 in my case) and so I complete the transfer (in the Linksys phones, that means hittint XFer again). The caller now, instead of being parked, disconnects. In the asterisk CLI it looks like this: ## the remote DUNDi user goes through some stuff and eventually dials to my local SIP extension: -- Executing [EMAIL PROTECTED]:11] Dial(IAX2/192.118.54.135:4569-2, SIP/2006||L()) in new stack -- Called 2006 -- SIP/2006-009e9e10 is ringing -- SIP/2006-009e9e10 answered IAX2/192.118.54.135:4569-2 ## I'm putting the caller on hold while I start the transfer -- Started music on hold, class 'default', on IAX2/192.118.54.135:4569-2 ## dialling 910 -- Executing [EMAIL PROTECTED]:3] Goto(SIP/2006-009eccc0, local-priv-incoming|910|1) in new stack -- Goto (local-priv-incoming,910,1) -- Executing [EMAIL PROTECTED]:1] Goto(SIP/2006-009eccc0, parkedcalls|700|1) in new stack -- Goto (parkedcalls,700,1) -- Executing [EMAIL PROTECTED]:1] Park(SIP/2006-009eccc0, ) in new stack == Parked SIP/2006-009eccc0 on [EMAIL PROTECTED] Will timeout back to extension [parkedcalls] s, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls # this I'm hearing -- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on SIP/2006-009eccc0 # now I complete the transfer == Spawn extension (parkedcalls, s, 1) exited KEEPALIVE on 'SIP/2006-009eccc0' -- Stopped music on hold on IAX2/192.118.54.135:4569-2 [Jan 15 15:22:17] WARNING[10582]: chan_sip.c:12310 handle_response: Notify answer on an owned channel? == Spawn extension (dundi-priv-lookup, 2006, 11) exited non-zero on 'IAX2/192.118.54.135:4569-2' -- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/192.118.54.135:4569-2, -- Done with call --) in new stack -- Hungup 'IAX2/192.118.54.135:4569-2' -- Stopped music on hold on SIP/2006-009eccc0 == SIP/2006-009eccc0 got tired of being parked # and this is where the remote caller disconnects Can you please tell me what I'm missing ? -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. The gates in my computer are AND, OR and NOT; they are not Bill. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk 1.4 Documentation
Cory Hawkless wrote: Is there an official list anywhere specifying the Prerequisites for installing asterisk(Specifally 1.4) on Fedora Core 4? I have been struggling with a “/configure: error: termcap support not found” /error when compiling 1.4 on my brand new install of FC4 fully updated, Fedora was installed as a base install to try and keep the overheads down of running a full GUI and all the other junk, but was lacking the ncurses-devel package. Once all of the prereq’s were installed it compiled fine, its frustrating I cant find a “This is what you must have installed before beginning your Asterisk install’ There are a lot of linux distros besides fedora and Asterisk 1.4 hasn't been out that long. You could take the knowledge you gained and contribute it to the wiki so others don't have to experience the same frustration. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MFC/R2 problems
Hi, thanks for the help. It turns out the this device I had, an Orion GSM gateway, does not talk MFC/R2, but some variant of R2, according to Steve U. thanks anyways :) Facundo Ameal wrote: These are the different meanings for the diferrent error codes: T1 TIMEOUT = 32769 T2 TIMEOUT = 32770 T3 TIMEOUT = 32771 UNEXPECTED MF SIGNAL= 32772 UNEXPECTED CAS = 32773 INVALID STATE = 32774 SET_CAS FAILURE = 32775 SEIZE ACK TIMEOUT = 32776 DEVICE IO ERROR = 32777 T1B TIMEOUT = 32778 I hope it helps. Greets On 1/8/07, yusuf [EMAIL PROTECTED] wrote: Hi, if that means I should post my config, here goes: zaptel: span=1,1,3,cas,hdb3,crc4 cas=1-15:1101 cas=17-31:1101 unicall.conf: protocolvariant=id,10,10 protocolend=cpe group=1 channel = 1-15 channel = 17-31 wanpipe1.conf FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CAS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO Josué Conti wrote: Hi Yusuf, how are you? It orders in the list its configurations, so that let us can help. Best Regards Josue 2007/1/8, yusuf [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Jan 8 13:04:11 DEBUG[12252]: chan_unicall.c:955 unicall_disable_ec: disabled echo cancellation on channel 1 Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] -- Hungup 'UniCall/1-1' What does - Unicall/1 protocol error. Cause 32769 mean, and can anyone help me. -- -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] EM ?
When I send a call from my TE410P using EM, the legacy PBX answers the call but doesn't route it. Any suggestions on what config settings to muck with? Do you have PRI ISDN or inband signaling trunks? Either way, it would be zapata.conf configs that would be the issue. zapata.conf [trunkgroups] spanmap = 1,1,1 spanmap = 3,2,3 [channels] switchtype=5ess signaling=em_w If it is inband signaling trunks, have you tried to use 'em' instead of 'em_w'? The 'w' is telling asterisk to wait for a wink before sending the digits. Perhaps the legacy PBX is not sending the wink. Also, I believe (but may be wrong) that the 'trunksgroups' section is for ISDN. Switchtype is only for PRI ISDN so is not needed for inband signaling. Don Pobanz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what happened to sip list peers
All, I had used 1.4beta3 for some time. I read all the changes etc... One of the changes was Sip show peers was changed to sip list peers. I changed my interface to accomidate that... Over the weekend I installed 1.4.0 release. It seems as though the sip list peers is GONE and now it is back to Sip show peers. Am I missing something? Why did it revert back? Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP transfer issue
Wondering if anyone on here can help with a niggling issue: One of our extensions is unable to make attended transfers at all. The phone in question is an Elmeg ip290, and works fine for direct transfers. However, on attempting to make an attended transfer, the first leg succeeds (the inbound call is placed on hold and gets MoH, the Elmeg user announces the call to the target extension), but upon completing the transfer, both parties get MoH, not each other. There is an entry in the asterisk logs as follows: chan_sip.c:6930 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]'. Both legs must reside on Asterisk box to transfer at this time. The incoming call, the Elmeg and the target extension are all on the same asterisk box. The Elmeg is behind NAT, but canreinvite=no and nat=yes are both set in the appropriate sip.conf sections for both the Elmeg and the target destination. Can anyone shed any light on this? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what happened to sip list peers
Jerry Geis wrote: Why did it revert back? The developer community (with input from a lot of users) decided the change was not the right thing to do, and it got changed back. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rt db lookup
On 1/15/07, Tim Connolly [EMAIL PROTECTED] wrote: Which command effects whether or not the * server will lookup a peer from the db even though the phone isn't registered locally? I have several * servers but I want any server to be able to lookup and send a call to phones registered on another server (SIP cluster?). You may want to look at DUNDi for this. http://www.dundi.info/ regards David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone
On Fri, Jan 12, 2007 at 02:33:54PM -0500, Doug Crompton wrote: I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa) I have used newer firmwares but find that 3.1.3 had less echo problems. Thanks again Doug for that detailed explanation. As for the DTMF playback level and DTMF playback length settings, what do you use? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
chester c young wrote:
cannot make Dial(...,,g) work correctly, although Dial(...,,gh) works
just fine. (to make matters worse, it does seem to work sometimes,
although once working or not working between changes it either works or
doesn't work all the time.)
g option to Dial only continues the dialplan if the destination
(called) leg of the call hangs up. It will NOT cause the dialplan to
continue if the source (calling) leg of the call hangs up.
When the calling channel hangs up, Asterisk will send the remaining leg
of the call to exten = h.
My paypal address is [EMAIL PROTECTED]
Example
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1},,g)
exten = _91NXXNXX,2,Noop(DESTINATION HANGUP)
exten = h,1,Noop(SOURCE HANGUP)
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Re: [asterisk-users] cepstral voice still nags after registration
Thanks Paul. I think it was nagging because the phpagi code looks to see if there is already a wav file before creating a new one. Since I had old ones with the nagging, it didn't create new ones. The problem I am having now is that it won't play it at all, just beeps. Thanks! On 1/12/07, Paul [EMAIL PROTECTED] wrote: blackwater dev wrote: I'm using trixbox and the asterisk agi. I downloaded a cepstral voice and worked with it until I got the code to do what I wanted. I then registered the voice today to get rid of the 'this voice is not yet registered, stuff yet it still does that. Any ideas on how to fix this? It told me my info was valid. Thanks! I am not using trixbox and I installed swift in /opt In my case the file of interest is: /opt/swift/voices/Diane/license.txt The file contains my name, company and a license key See if you have a license.txt file like that in the right place Set it to root:root ownership and 644 permissions ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
On Monday 15 January 2007 11:03 am, Eric ManxPower Wieling wrote: g option to Dial only continues the dialplan if the destination (called) leg of the call hangs up. It will NOT cause the dialplan to continue if the source (calling) leg of the call hangs up. I was going to give him the exact same answer, but he specifically said it's not going on when the called party hangs up. I'm using 'g' just fine and it works exactly as you describe, so I'm guessing that something else is the case. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ANY ADVICE ON THIS????
Hello List, I am stuck with this problem for several days... anybody can give me a hint on this?? I know many of you dealt with problems similar to this, how did you address this?? Thanks in advance!!! -lars -- Forwarded message -- From: Lars Knopf [EMAIL PROTECTED] Date: Jan 11, 2007 1:12 PM Subject: realtime sipusers and rtcachefriends... big headache!! To: asterisk-users@lists.digium.com hi folks, I am using asterisk 1.2.13 (debian etch). My customer's sip accounts are stored in realtime sipusers. I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes Each account has nat=yes Now, I have lot of problems. for example, when I change the 'secret' field of a user in the database, it doesn't get reflected in Asterisk, who is still expecting the old password. Randomly, when trying to dial SIP/x (a customer's account), especially those behind NAT, I get in the console the error no route to Sometimes, too, they can't even register with asterisk. It seems to happen mostly when using realtime. I was digging into the bug tracking system, and I see two bugs that seems to be related, but I can't figure how can I fix it or what step I am supposed to do. The bugs are: http://bugs.digium.com/view.php?id=4687 http://bugs.digium.com/view.php?id=4832 So please, anything than can bring me some light on this... is very appreciated. -lars ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ANY ADVICE ON THIS????
On 1/15/07, Lars Knopf [EMAIL PROTECTED] wrote: Hello List, I am stuck with this problem for several days... anybody can give me a hint on this?? I know many of you dealt with problems similar to this, how did you address this?? Thanks in advance!!! -lars -- Forwarded message -- From: Lars Knopf [EMAIL PROTECTED] Date: Jan 11, 2007 1:12 PM Subject: realtime sipusers and rtcachefriends... big headache!! To: asterisk-users@lists.digium.com hi folks, I am using asterisk 1.2.13 (debian etch). My customer's sip accounts are stored in realtime sipusers. I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes Each account has nat=yes Now, I have lot of problems. for example, when I change the 'secret' field of a user in the database, it doesn't get reflected in Asterisk, who is still expecting the old password. Randomly, when trying to dial SIP/x (a customer's account), especially those behind NAT, I get in the console the error no route to Sometimes, too, they can't even register with asterisk. It seems to happen mostly when using realtime. I was digging into the bug tracking system, and I see two bugs that seems to be related, but I can't figure how can I fix it or what step I am supposed to do. The bugs are: http://bugs.digium.com/view.php?id=4687 http://bugs.digium.com/view.php?id=4832 So please, anything than can bring me some light on this... is very appreciated. I think you will need to prune the user/peer after changes. I believe the syntax is something like sip prune realtime user_or_peer where user_or_peer is the actual username. - David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] php agi - first phrase truncated, all others fine
I have the following code. When I call the extension, it either ignores the
first Hello there everyone, or says hello and moves on sometime stoping
before it finishes hello. The rest of the text reads fine. Anyone else
have this issue??
Thanks!
require('/var/lib/asterisk/agi-bin/phpagi.php');
$agi = new AGI();
$agi-answer();
$agi-swift(Hello there everyone );
$agi-swift(Please press 1 for a search .);
$result= $agi-get_data('beep',3, 1);
$zip= $result['result'];
$agi-swift(That concludes your call. Thank you, Good bye .);
$agi-hangup();
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[asterisk-users] .call files - no hangup
hi all, i have the following .call file: Channel: IAX2/[EMAIL PROTECTED]/myPOTSline MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: default Extension: 156 Priority: 1 when i drop the .call file into the /var/spool/asterisk/outgoing/ it calls out on voipjet, connects to extension 156 (which runs the a2billing AGI) and everything is great - except that if i hang up the PSTN side, nothing happens. Only when the AGI decides to hang up does it hang up. Just for reference, extension 156 in default is: exten = 156,1,Answer exten = 156,2,Wait,1 exten = 156,3,DeadAGI(a2billing.php) exten = 156,4,Hangup anyone have any idea why a hang up on the PSTN side is not being accepted? thanks, yair ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Has been working for 9 Months - Very Very StrangeI cannot dial specific extensions from my dialplan - NOT ACONTEXT PROBLEM!!
with tcpdump i could notice that invites didn't reach my * server. After Rebooting Lan's Firewall CheckPoint problem solved. On 1/12/07, Steven [EMAIL PROTECTED] wrote: Is there a local dialplan on the phone? Maybe these phones were recently upgraded or reset to factory and lost the 4XXX dialplan. That is where I would start. -- -- Steven http://www.glimasoutheast.org Marco Mouta [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX - BROKEN but not for every account. Also I notice that for SIP accounts that can't dial 4XXX they can dial *98 and PSTN calls, and yes they are all in the same context since April 2006! SIP to PSTN - OK SIP to IAX - OK This is a graph from ethereal: Dialing 4214, my own SIP extension! |Time | 192.168.34.26 | XXX.XXX.XX.XX | |11,219 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |11,721 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |12,727 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |14,739 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |18,762 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | Dialing *98 to check voicemail: 2|21,882 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | 2|21,884 | 407 Proxy Authentication Required |SIP Status | |(2752) -- (61414) | 2|21,886 | ACK | |SIP Request | |(2752) -- (5060) | 2|21,990 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | 2|21,991 | 100 Trying| |SIP Status | |(2752) -- (61414) | 2|21,997 | 200 OK SDP ( g711A GSM g711U telephone-event) |SIP Status | |(2752) -- (61414) | 2|22,034 | RTP (g711U) |RTP Num packets:116 Duration: 2.315s ssrc:490185229 | |(42576) -- (18670) | 2|22,208 | ACK | |SIP Request | |(2752) -- (5060) | 2|23,025 | RTP (g711U) |RTP Num packets:75 Duration:1.484s ssrc:1496378340 | |(42576) -- (18670) | 2|24,523 | BYE | |SIP Request | |(2752) -- (5060) | 2|24,525 | 200 OK| |SIP Status | |(61413) -- (5060) | 2|25,026 | BYE | |SIP Request | |(2752) -- (5060) | 2|25,027 | 200 OK| |SIP Status | |(61413) -- (5060) | Also I notice, with SIP debug peer 4214 on * CLI , that when i dial from my sip phone 4XXX numbers, nothing seems to reach the asterisk Server! I hope someone can point me out where is the problem! This server has only sip extensions. P4 - 1G RAM wiht TE110P with weekly reboot. Best regards, Marco Mouta -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] php agi - first phrase truncated, all others fine
Are you creating a temporary wav file? If so, look at that first. If the
wav file is truncated at least you know the problem is related to the
way swift gets invoked and passed the text argument. If the file is okay
you need to look at the way it gets handled afterwards.
blackwater dev wrote:
I have the following code. When I call the extension, it either
ignores the first Hello there everyone, or says hello and moves on
sometime stoping before it finishes hello. The rest of the text reads
fine. Anyone else have this issue??
Thanks!
require('/var/lib/asterisk/agi-bin/phpagi.php');
$agi = new AGI();
$agi-answer();
$agi-swift(Hello there everyone );
$agi-swift(Please press 1 for a search .);
$result= $agi-get_data('beep',3, 1);
$zip= $result['result'];
$agi-swift(That concludes your call. Thank you, Good bye .);
$agi-hangup();
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Re: [asterisk-users] Stumped with Dial - $50 for answer
g option to Dial only continues the dialplan if the destination (called) leg of the call hangs up. It will NOT cause the dialplan to continue if the source (calling) leg of the call hangs up. When the calling channel hangs up, Asterisk will send the remaining leg of the call to exten = h. this is exactly right and is exactly the problem. when the called leg hangs up the dial plan does not proceed to the next priority. Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P, fxotune and ADSL filters - Just a FYI, FWIW
This may be commonly known but I haven't come across it so here goes, maybe it'll help someone: I have terrible echo with asterisk 1.2, zaptel 1.2.12, and a TDM400P with 1 FXS and two FXO modules. The Mark2 echo canceller with Aggressive turned on was the only setting that would make it acceptable. I found fxotune with this zaptel version to be broken. I pulled the latest fxotune.c and fxotune.h from cvs and recompiled zaptel. fxotune then ran but I got the error: Could not fill input buffer - got -1 bytes, expected 4000 bytes Failure! After two days I installed a splitter to listen in and found out that fxotune wanted 18 seconds of silence on the line but Bellsouth only gives 15 seconds. The -m switch in ./fxotune -i -m 15 -vv -b 3 -e 4 -t 2 allowed the test to complete successfully. Before tuning the TDM400P with ./fxotune -s, I observed the echo percentage on the line with ./fxotune -d -b 4 to be .32, a far cry from the .05 I wanted. After ./fxotune -s, ./fxotune -d -b 4 revealed an echo percentage of .075, still not good enough. I remembered that there is a DSL filter between this FXO module and the PSTN to break out signal for my DSL modem. I removed it and plugged the FXO straight in to PSTN. After a rerun of ./fxotune -i -m 15 -vv -b 3 -e 4 -t 2 and a ./fxotune -s, ./fxotune -d -b 4 now reveals .026 percent echo! It appears that the DSL filter circuitry affects the .fxotune impedance test to the point that it becomes ineffective (~.05 delta in my case) FWIW, I replaced the filter and reran ./fxotune -d -b 4 and observed a report of .11 percent echo, which I do not trust due to the filter's effect on the circuit. I eagerly removed the aggressive suppression and restored the original echo canceller to be disappointed that the echo still exists. So it is back to Mark2 with Aggressive. If you hang a FXO module behind a DSL filter and have high echo percentages or echo, this is a gotcha. I'm now experimenting with zaptel 1.4 with similar results, despite a new default echo algorithm. Also, any tips on echo reduction from here would be greatly appreciated, I'm out of ideas. My biggest fear is installing a hybrid system in a client's office and to come across a situation where I can't suppress echo.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I have to register asterisk/sip with a sipproxy that does not support authentication?
I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request? # sip.conf [general] insecure=very permit=207.148.115.10/255.255.255.0 [myproxy] type=friend host=217.118.115.10 context=from-sip # Logging: --- Reliably Transmitting (NAT) to 207.148.115.10:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 207.148.115.10:5060;branch=z9hG4bK3c4c865c4861d0ec0dc19fa40406cdf4;received=207.148.115.10 From: sipx.at-n.com sip:[EMAIL PROTECTED]:5060;tag=as1cc62bc2 To: sip:[EMAIL PROTECTED];tag=as6b72831a Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7b7f2a63 Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) Found user 'julien' sip*CLI --- SIP read from 207.148.115.10:5060 --- ACK sip:207.148.115.20 SIP/2.0 Max-Forwards: 69 From: sipx.at-n.com sip:[EMAIL PROTECTED]:5060;tag=as1cc62bc2 To: sip:[EMAIL PROTECTED];tag=as6b72831a Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Via: SIP/2.0/UDP 207.148.115.10:5060;branch=z9hG4bK3c4c865c4861d0ec0dc19fa40406cdf4 Content-Length: 0 - --- (8 headers 0 lines) --- sip*CLI --- SIP read from 207.148.115.10:5060 --- SIP/2.0 407 Proxy Authentication Required Proxy-Authenticate: Digest algorithm=MD5,realm=asterisk,nonce=7b7f2a63 To: sip:[EMAIL PROTECTED];tag=3377872210-792296 From: sipx.at-n.com sip:[EMAIL PROTECTED];tag=as1cc62bc2 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 207.148.115.20:5060;branch=z9hG4bK613354f9;rport Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone
Doug, You are saying that RFC2833 somehow doesn't work if you have the Asterisk AND at a distinct time (still within the same call), the callee to see the DTMF, correct ? Would this be in any case ? (meaning, if the voice path is going via the Asterisk or UA to UA directly ?) I've my spa3k right now somewhat far :-), and I can't test it, but you know by any chance if SIP INFO would suffer from the same curse :-) ? From my limited understand, a big difference in this case is that RFC2833 really is in the RTP stream, but is not voice payload, while with SIP INFO, is done 100% out-of-band. Doug Crompton wrote: I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa) I have used newer firmwares but find that 3.1.3 had less echo problems. Connect a real analog phone to spa3000 fxs. Call it from another source, when connected send DTMF tones from that source. You should hear at least 100ms or more of the tone. inband should work. I suspect you are using alaw or ulaw codecs. There is really no reason to use anything else. When it does not work you will hear nothing more then a click or an ocassional to short tone. Another thing to check is that you should not be using any transfer options in your dial statement (t or T or other special features. You really have to listen to this to check it and make changes. Be sure to restart both spa3000 and asterisk when you make changes. Otherwise you can get fooled. If you are making the call from the spa3000 fxo to fxs, you need to have inband in BOTH. This is a known bug in Asteriskspa3000 for dtmf. I think the problem is somewhat shared but improvements in 1.4 may gelp or fic the problem. I am using 1.2 so I cannot answer that. Basically when using the spa3000 you have to make the choice of wether you want to be able to use dtmf features (transfer etc.)OR have the capability to send DTMF to or from the caller or callee. you really can't have both. Thus inband vs. rfc2833. I chose inband so I can interact with called ivr's and call in from pstn and access my VM. Doug On Fri, 12 Jan 2007, Louis-David Mitterrand wrote: On Fri, Jan 12, 2007 at 10:58:16AM -0500, Doug Crompton wrote: The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling. Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be the line1 tab on spa3000. This applies to the fxo (pstn) also if you are using it for such things as ivr's. Thanks for your suggestion. We tried that without success (using firmware 3.1.7(GWc)) Do you think an upgrade to 3.1.10 might be warranted? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I have to register asterisk/sip with a sipproxy that does not support authentication?
Julien Chavanton wrote: I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request? SIP clients never request authentication/authorization. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue and Interface time out
We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone
I use default values for both of those. The big thing is to call youself. Use a cell, call a phone on the FXS. Hit a key on the cell and listen on FXS for DTMF. Make changes, reboot, and repeat. Hearing is believing. It is so much easier! I think you will find the inband will work. Doug On Mon, 15 Jan 2007, Louis-David Mitterrand wrote: On Fri, Jan 12, 2007 at 02:33:54PM -0500, Doug Crompton wrote: I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa) I have used newer firmwares but find that 3.1.3 had less echo problems. Thanks again Doug for that detailed explanation. As for the DTMF playback level and DTMF playback length settings, what do you use? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue and Interface time out
try autopause in queues.conf James Fromm wrote: We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone
I am not sure what you are asking? The problem is that rfc2833 does not play well with the spa-3000 and Asterisk. I am not sure if it is limited to just the spa3k. There is a bug causing this that has been documented. Google spa3000 dtmf bug asterisk for more info. The bottom line is that you need to use sip info (inband dtmf) if you desire dtmf transfer to the other party after the call has completed. Such as you call a bank, or you call your Asterisk voicemail, or your door lock which is actuated by dtmf. If none of these are of interest and you would rather have the dtmf features of Asterisk, then use rfc2833. you can't have both! Doug On Mon, 15 Jan 2007, Julio Arruda wrote: Doug, You are saying that RFC2833 somehow doesn't work if you have the Asterisk AND at a distinct time (still within the same call), the callee to see the DTMF, correct ? Would this be in any case ? (meaning, if the voice path is going via the Asterisk or UA to UA directly ?) I've my spa3k right now somewhat far :-), and I can't test it, but you know by any chance if SIP INFO would suffer from the same curse :-) ? From my limited understand, a big difference in this case is that RFC2833 really is in the RTP stream, but is not voice payload, while with SIP INFO, is done 100% out-of-band. Doug Crompton wrote: I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa) I have used newer firmwares but find that 3.1.3 had less echo problems. Connect a real analog phone to spa3000 fxs. Call it from another source, when connected send DTMF tones from that source. You should hear at least 100ms or more of the tone. inband should work. I suspect you are using alaw or ulaw codecs. There is really no reason to use anything else. When it does not work you will hear nothing more then a click or an ocassional to short tone. Another thing to check is that you should not be using any transfer options in your dial statement (t or T or other special features. You really have to listen to this to check it and make changes. Be sure to restart both spa3000 and asterisk when you make changes. Otherwise you can get fooled. If you are making the call from the spa3000 fxo to fxs, you need to have inband in BOTH. This is a known bug in Asteriskspa3000 for dtmf. I think the problem is somewhat shared but improvements in 1.4 may gelp or fic the problem. I am using 1.2 so I cannot answer that. Basically when using the spa3000 you have to make the choice of wether you want to be able to use dtmf features (transfer etc.)OR have the capability to send DTMF to or from the caller or callee. you really can't have both. Thus inband vs. rfc2833. I chose inband so I can interact with called ivr's and call in from pstn and access my VM. Doug On Fri, 12 Jan 2007, Louis-David Mitterrand wrote: On Fri, Jan 12, 2007 at 10:58:16AM -0500, Doug Crompton wrote: The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling. Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be the line1 tab on spa3000. This applies to the fxo (pstn) also if you are using it for such things as ivr's. Thanks for your suggestion. We tried that without success (using firmware 3.1.7(GWc)) Do you think an upgrade to 3.1.10 might be warranted? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
On 1/15/07, chester c young [EMAIL PROTECTED] wrote: g option to Dial only continues the dialplan if the destination (called) leg of the call hangs up. It will NOT cause the dialplan to continue if the source (calling) leg of the call hangs up. When the calling channel hangs up, Asterisk will send the remaining leg of the call to exten = h. this is exactly right and is exactly the problem. when the called leg hangs up the dial plan does not proceed to the next priority. Silly question: how are the calls going out? If they're going out through an analog line without the ability to detect hang-ups, then, well, that's the problem. We have this with a few of our TDM400's, as well as an old X100P. callprogress=yes did not seem to fix them much. So, the result is that our phone system always thinks we are the ones hanging up. Sometimes that causes a bit of a problem when a person is in a queue and hangs up before they get to an agent. In those cases, the agent gets the dead line. But, when they hang up, the line is freed. In that case, you would just have to use the 'h' flag, and put the rules there, and realize that your system will always believe you hung up. The other option is to get a line with disconnect supervision from your phone company, or some type of digital trunk (PRI, etc). Hope that helps, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM400P, fxotune and ADSL filters - Just a FYI, FWIW
Eureka, echo free at last! ahh I set the rxgain by running my CO's milliwatt test to 14844 from the original 6688. I just looped from FXO 3 to the asterisk milliwatt() test on FXO 4, I just found the txgain was 6686, instead of 14844. (http://lists.digium.com/pipermail/asterisk-users/2004-November/064312.h tml) After bumping the txgain to 6 (!), I got it to 13500 and that was all I could get. However, The echo has disappeared. Sorry to answer a question that hasn't been asked, but maybe this will save someone some serious frustration! _ From: John French Sent: Monday, January 15, 2007 11:40 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM400P, fxotune and ADSL filters - Just a FYI, FWIW This may be commonly known but I haven't come across it so here goes, maybe it'll help someone: I have terrible echo with asterisk 1.2, zaptel 1.2.12, and a TDM400P with 1 FXS and two FXO modules. The Mark2 echo canceller with Aggressive turned on was the only setting that would make it acceptable. I found fxotune with this zaptel version to be broken. I pulled the latest fxotune.c and fxotune.h from cvs and recompiled zaptel. fxotune then ran but I got the error: Could not fill input buffer - got -1 bytes, expected 4000 bytes Failure! After two days I installed a splitter to listen in and found out that fxotune wanted 18 seconds of silence on the line but Bellsouth only gives 15 seconds. The -m switch in ./fxotune -i -m 15 -vv -b 3 -e 4 -t 2 allowed the test to complete successfully. Before tuning the TDM400P with ./fxotune -s, I observed the echo percentage on the line with ./fxotune -d -b 4 to be .32, a far cry from the .05 I wanted. After ./fxotune -s, ./fxotune -d -b 4 revealed an echo percentage of .075, still not good enough. I remembered that there is a DSL filter between this FXO module and the PSTN to break out signal for my DSL modem. I removed it and plugged the FXO straight in to PSTN. After a rerun of ./fxotune -i -m 15 -vv -b 3 -e 4 -t 2 and a ./fxotune -s, ./fxotune -d -b 4 now reveals .026 percent echo! It appears that the DSL filter circuitry affects the .fxotune impedance test to the point that it becomes ineffective (~.05 delta in my case) FWIW, I replaced the filter and reran ./fxotune -d -b 4 and observed a report of .11 percent echo, which I do not trust due to the filter's effect on the circuit. I eagerly removed the aggressive suppression and restored the original echo canceller to be disappointed that the echo still exists. So it is back to Mark2 with Aggressive. If you hang a FXO module behind a DSL filter and have high echo percentages or echo, this is a gotcha. I'm now experimenting with zaptel 1.4 with similar results, despite a new default echo algorithm. Also, any tips on echo reduction from here would be greatly appreciated, I'm out of ideas. My biggest fear is installing a hybrid system in a client's office and to come across a situation where I can't suppress echo.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Software callcenter
Hello everybody Anyone know a software for callcenter, with statistics and reports and work with asterisk? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue and Interface time out
NICE! That did the trick. Thanks! Julian Lyndon-Smith wrote: try autopause in queues.conf James Fromm wrote: We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Software callcenter
Hello, There are two GPL call center suites that handle inbound and outbound calling for Asterisk: VICIDIAL: http://astguiclient.sf.net/vicidial.html GnuDialer: http://www.gnudialer.org MATT--- On 1/15/07, Carlos Rojas [EMAIL PROTECTED] wrote: Hello everybody Anyone know a software for callcenter, with statistics and reports and work with asterisk? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Console latency
From:"Yuan LIU" [EMAIL PROTECTED]Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms).I can run the same application from a GrandStream phone (on the same LAN) and hear little delay.What could possibly be wrong?If it were interrupt overload, I'd hear lots of cracks in my echo, right?I'm not hearing that.Besides, a telephony card is not involved. For future reference, Ihave determinedthat this was related to theISA card used. It may not be entirely due to card, but when I replaced the card with a newer card (a SIIG Wavetable 5.1 PCI, identified as CMI8738-MC6 ), things improved significantly. Yuan Liu I'm running asterisk-1.2.13andzaptel-1.2.10 on Linux 2.6.15-27-386 (Ubuntu 6 distribution without X).Hardware includes a P III 600 MHz, 386 MB RAM, an X100P card that's not part of this test (also used an X100P clone card to same result), and a CS4239 sound card (ISA) with ALSA driver (also tried with OSS to similar result but OSS had a harder time getting volume up).ALSA needed a bit of tweak to work properly with CS4239, but afer carefully setting alsamixer, I don't hear much echo when making calls from the console.Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] php agi - first phrase truncated, all others fine
Yes, the wav file is fine. For some reason it's just getting cut off.
Whatever I type there seems to get cut off, strange.
On 1/15/07, Paul [EMAIL PROTECTED] wrote:
Are you creating a temporary wav file? If so, look at that first. If the
wav file is truncated at least you know the problem is related to the
way swift gets invoked and passed the text argument. If the file is okay
you need to look at the way it gets handled afterwards.
blackwater dev wrote:
I have the following code. When I call the extension, it either
ignores the first Hello there everyone, or says hello and moves on
sometime stoping before it finishes hello. The rest of the text reads
fine. Anyone else have this issue??
Thanks!
require('/var/lib/asterisk/agi-bin/phpagi.php');
$agi = new AGI();
$agi-answer();
$agi-swift(Hello there everyone );
$agi-swift(Please press 1 for a search .);
$result= $agi-get_data('beep',3, 1);
$zip= $result['result'];
$agi-swift(That concludes your call. Thank you, Good bye .);
$agi-hangup();
___
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[asterisk-users] Delay in Call Distribution using the Queue Application
Hello all, we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue application. If there are many calls in the queue, it sometimes takes up to 30 Seconds before a call is distributed to an agent. For example there are 10 callers in the queue, an Agent is finishing a call and it takes up to 30 seconds before his phone rings again. We're already set the wrapuptime parameter in queues.conf to 0, for my point of view an agent phone that becomes available again should ring immediately after hanging up a call. Does anybody know if there are any known issues or restrictions in the queue application in version 1.2.12.1? Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nufone
Are these guys still around? I cannot get to www.nufone.net or nufone.com Thanks, Wiley ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help create asterisk cookbook
I have not yet seen this article posted to this list, so I thought many of us would be interested in having a look at this project sponsored by O'Reilly: http://www.oreillynet.com/etel/blog/2007/01/help_create_the_asterisk_cookb.html It seems they are looking for both problems and solutions, and I'm sure we'll have plenty :) l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Software callcenter
We offer a commecial very detailed reporting solution that is widely deployed and is available free of charge to small CCs / SOHOs. See http://queuemetrics.com . Which kind of call center are you going to implement? inbound / outbound / mixed traffic? l. On Mon, 15 Jan 2007 19:36:11 +0100, Carlos Rojas [EMAIL PROTECTED] wrote: Hello everybody Anyone know a software for callcenter, with statistics and reports and work with asterisk? Regards -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PBX '' '||' Grandstream GXP-2000 problem
Hi People, We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHz Box... The issues that we are experiencing involves our Telephone Operator's/Receptionist whom answer multiple incoming calls... As an example.., when they answer line 1 and Line 2 starts to ring they would ask the person on line 1 to hold and proceed to answer line 2 and forward line 2 to to the requested extension. The problem is when they attempt to pick line 1 off the hold in order to handle that call, line 1 is either dropped or the Grandstream Phone freezes and the user is forced to rest the phone. The situation persist whenever there are multiple lines active with incoming calls and upon answering one, placing the line on hold and attempting to answer the other lines active calls will be dropped the the phone just hangs/freezes. We know that the call is dropped because the people call back complaining about being hung up on We have had our dedicated T1 (for voice only) tested several times and it is good. We have had the Asterisk PBX completely redone and gone over thoroughly and are at the point where we are suspecting the configuration file for the Grandstream GXP-2000 Telephone as the culprit. We would like to know what suggestions anyone out there might have if any... Thanks, Jose P. Espinal DomiNET ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nufone
I can connect to http://www.nufone.net/ just fine. Wiley Siler wrote: Are these guys still around? I cannot get to _www.nufone.net_ file://www.nufone.net or nufone.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue cmd option 'i'
Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should ignore call forward requests from queue members and do nothing when they are requested. Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding should be ignored. Using Queue(customerservice|i) causes Asterisk to crash when sending the call to the member with forwarding enabled on their SIP device. Am I misinterpreting what this option does? Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Software callcenter
On Mon, 2007-01-15 at 20:37 +0100, Lenz wrote: [..] Hello everybody Anyone know a software for callcenter, with statistics and reports and work with asterisk? Try MOR from www.kolmisoft.com Regards, Regards -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] php agi - first phrase truncated, all others fine
Looking at the phpagi 2.14 source that I have I see that right after
creating the file it does this:
$ret = $this-stream_file($fname, $escape_digits);
So if the swift-generated wav file sounds right the stream_file is where
the problem lies.
copy the wav file to a file named test.wav and create an extension to
play it using the phpagi library. That will show you if the stream_file
function is breaking or not.
blackwater dev wrote:
Yes, the wav file is fine. For some reason it's just getting cut
off. Whatever I type there seems to get cut off, strange.
On 1/15/07, *Paul* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Are you creating a temporary wav file? If so, look at that first.
If the
wav file is truncated at least you know the problem is related to the
way swift gets invoked and passed the text argument. If the file
is okay
you need to look at the way it gets handled afterwards.
blackwater dev wrote:
I have the following code. When I call the extension, it either
ignores the first Hello there everyone, or says hello and
moves on
sometime stoping before it finishes hello. The rest of the text
reads
fine. Anyone else have this issue??
Thanks!
require('/var/lib/asterisk/agi-bin/phpagi.php');
$agi = new AGI();
$agi-answer();
$agi-swift(Hello there everyone );
$agi-swift(Please press 1 for a search .);
$result= $agi-get_data('beep',3, 1);
$zip= $result['result'];
$agi-swift(That concludes your call. Thank you, Good bye .);
$agi-hangup();
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Re: [asterisk-users] Nufone
I second that, seems to be working fine from here (Toronto/Rogers fiber connection). Maybe a lagging DNS or routing issue with your ISP? Alex On 1/15/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: I can connect to http://www.nufone.net/ just fine. Wiley Siler wrote: Are these guys still around? I cannot get to _www.nufone.net_ file://www.nufone.net or nufone.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX '' '||' Grandstream GXP-2000 problem
On 1/15/07, J. Espinal [EMAIL PROTECTED] wrote: Hi People, We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHzBox... The issues that we are experiencing involves our Telephone Operator's/Receptionist whom answer multiple incoming calls... As an example.., when they answer line 1 and Line 2 starts to ring they would ask the person on line 1 to hold and proceed to answer line 2 and forward line 2 to to the requested extension. The problem is when they attempt to pick line 1 off the hold in order to handle that call, line 1 is either dropped or the Grandstream Phone freezes and the user is forced to rest the phone. The situation persist whenever there are multiple lines active with incoming calls and upon answering one, placing the line on hold and attempting to answer the other lines active calls will be dropped the the phone just hangs/freezes. We know that the call is dropped because the people call back complaining about being hung up on We have had our dedicated T1 (for voice only) tested several times and it is good. We have had the Asterisk PBX completely redone and gone over thoroughly and are at the point where we are suspecting the configuration file for the Grandstream GXP-2000 Telephone as the culprit. We would like to know what suggestions anyone out there might have if any... Thanks, Are you using G.729? Last I heard, grandstreams could only have one call via G.729 at a time. It had something to do with the licensing that they used, I think. Just a thought... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read Voicmail Boxes
If you would bother to read my post you will see that what I am wanting to do is not the asterisk directory cmd. I don't want them to be able to search or anything fancy like that. I want an app that will go through and say the recorded name for everyone that has a mailbox one by one. I did search but was not able to find anything that can do what I want. As far as the English class goes; please be more careful about reading posts and trying to understand exactly what is being asked before you flame me for not searching. Thanks, _ /-\ ndrew On 1/12/07, Andrew Joakimsen [EMAIL PROTECTED] wrote: I think what you want is called a directory, no? I'm not positive because the English language isn't my main expertise, I know more about Linux and stuff like that. Maybe you can find a newsgroup about English and get an answer to that -- or better yet tell them to write all your other mailing list posting for you! But perhaps if you would have searched for directory on voip-info.org before you posted this message you would have found your answer. This is a user's discussion list not free for all tech support for people who don't know the meaning of search ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] php agi - first phrase truncated, all others fine
blackwater dev wrote:
I have the following code. When I call the extension, it either
ignores the first Hello there everyone, or says hello and moves on
sometime stoping before it finishes hello. The rest of the text reads
fine. Anyone else have this issue??
Try adding this...
require('/var/lib/asterisk/agi-bin/phpagi.php');
$agi = new AGI();
$agi-answer();
sleep(0.5);
$agi-swift(Hello there everyone );
Regards,
Trevor
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Re: [asterisk-users] php agi - first phrase truncated, all others fine
From:"blackwater dev" [EMAIL PROTECTED]
Yes, the wav file is fine. For some reason it's just getting cut off. Whatever I type there seems to get cut off, strange.
With plain dialplan (no AGI), I notice that the first few syllables from Playback() or Background()could be eaten up, sometimes more, sometimes less. A little Wait() after Answer()may help "stablize" the voice channel.
Yuan Liu
On 1/15/07, Paul [EMAIL PROTECTED] wrote:
Are you creating a temporary wav file? If so, look at that first. If the
wav file is truncated at least you know the problem is related to theway swift gets invoked and passed the text argument. If the file is okayyou need to look at the way it gets handled afterwards.blackwater dev wrote:
I have the following code.When I call the extension, it either ignores the first "Hello there everyone", or says "hello" and moves on sometime stoping before it finishes hello.The rest of the text reads
fine.Anyone else have this issue?? Thanks!require('/var/lib/asterisk/agi-bin/phpagi.php'); $agi = new AGI(); $agi-answer(); $agi-swift("Hello there everyone ");
$agi-swift("Please press 1 for asearch."); $result= $agi-get_data('beep',3, 1); $zip= $result['result'];
$agi-swift("That concludes your call.Thank you, Good bye ."); $agi-hangup();___
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[asterisk-users] Audiocodes Mediant 1000, Polycom, and no ringback on transfer
I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I am using Polycom 501's and 601',s and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, the PSTN party hears dead air until the call is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any one have any experience with this type of setup? Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone
Doug Crompton wrote: I am not sure what you are asking? The problem is that rfc2833 does not play well with the spa-3000 and Asterisk. I am not sure if it is limited to just the spa3k. There is a bug causing this that has been documented. Google spa3000 dtmf bug asterisk for more info. The bottom line is that you need to use sip info (inband dtmf) if you desire dtmf transfer to the other party after the call has completed. Such as you call a bank, or you call your Asterisk voicemail, or your door lock which is actuated by dtmf. If none of these are of interest and you would rather have the dtmf features of Asterisk, then use rfc2833. you can't have both! SIP INFO is not the same as Inband DTMF, that is why I'm asking. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nufone
Wiley Siler wrote: Are these guys still around? I cannot get to _www.nufone.net_ file://www.nufone.net or nufone.com Not only can I get to their website, but yesterday I called their customer service and for the first time ever it was actually answered by a live person. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay in Call Distribution using the Queue Application
[EMAIL PROTECTED] wrote: Hello all, we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue application. If there are many calls in the queue, it sometimes takes up to 30 Seconds before a call is distributed to an agent. For example there are 10 callers in the queue, an Agent is finishing a call and it takes up to 30 seconds before his phone rings again. We're already set the wrapuptime parameter in queues.conf to 0, for my point of view an agent phone that becomes available again should ring immediately after hanging up a call. Does anybody know if there are any known issues or restrictions in the queue application in version 1.2.12.1? Thanks and Regards Check out the red highlighted paragraph. Maybe that is connected. http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone
OK... I understand. As I remember I did try other methods like INFO. It has been awhile. I think INBAND is the only one that worked for me. Doug On Mon, 15 Jan 2007, Julio Arruda wrote: Doug Crompton wrote: I am not sure what you are asking? The problem is that rfc2833 does not play well with the spa-3000 and Asterisk. I am not sure if it is limited to just the spa3k. There is a bug causing this that has been documented. Google spa3000 dtmf bug asterisk for more info. The bottom line is that you need to use sip info (inband dtmf) if you desire dtmf transfer to the other party after the call has completed. Such as you call a bank, or you call your Asterisk voicemail, or your door lock which is actuated by dtmf. If none of these are of interest and you would rather have the dtmf features of Asterisk, then use rfc2833. you can't have both! SIP INFO is not the same as Inband DTMF, that is why I'm asking. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes Mediant 1000, Polycom, and no ringback on transfer
On 1/15/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I don't have any experience with an Audiocodes Meidant 1000, but I'll try to help you I am using Polycom 501's and 601',s We have a lot of these and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, I don't think you can do that. Here's why: on the Polycom's, the Transfer button doesn't reappear until the transferree picks up the phone. Unless something changed in the firmware recently. But, if you're completing it before the 3rd party answers, it's not an attended transfer. the PSTN party hears dead air until the call is answered or goes to voicemail. I would start by making sure the Music on Hold actually works, and that the SIP phones are properly configured to use a MOH context that actually exists. If those things check out, I would try using a blind transfer and see what happens, try transferring when the 3rd party answers (VM or whatever), and watch the console carefully with as much verbosity as possible. I'm not really sure where to start my troubleshooting. Any one have any experience with this type of setup? Hope this helps, David Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Nufone
Strange. I can get there too now... Must have been DNS problem Now to figure out where my DID has gone Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Prior Sent: Monday, January 15, 2007 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Nufone Wiley Siler wrote: Are these guys still around? I cannot get to _www.nufone.net_ file://www.nufone.net or nufone.com Not only can I get to their website, but yesterday I called their customer service and for the first time ever it was actually answered by a live person. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] connecting 2 asterisk servers through OpenVPN
I am trying to connect 2 asterisk servers through OpenVPN, the VPN should carry 16 channel, however when active channels reached 4 concurrent channels, the connection became unstable, with a very high latency (around 900ms), the internet bandwidth is 1Mbps on each server, I have upgraded the bandwidth to double it, but still have exactly the same problem. Any tips or recommendations on such setup? I am using SIP and G729 between the 2 servers, openVPN using UDP with no compression. Thanks, O.Youssef ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Addpac 2620 don't relay DTMF to PSTN
Hi Guys: I'm using Asterisk with Addpac 2620 as gateway, internally I'm using Grandstream BT200, unfortunately when I called to external phones (PSTN), and I have to choose some extensions, the Phone don't dial the extensions, I believe that DTMF relay in ADDPAC is not working well. I'm using RFC 2833 and ALaw for SIP Channel (Between ASterisk and ADDPAC). Someone have any experiencie with Addpac as gateway, or some workaround about this issue. Thanks! -- Omar E.P.T - Certified Networking Professionals make better Connections! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay in Call Distribution using the Queue Application
On Monday 15 January 2007 19:22, [EMAIL PROTECTED] wrote: Hello all, For example there are 10 callers in the queue, an Agent is finishing a call and it takes up to 30 seconds before his phone rings again. We're already set the wrapuptime parameter in queues.conf to 0, for my point of view an agent phone that becomes available again should ring immediately after hanging up a call. Try setting wrapuptime to 1 . Setting it to zero likely enables some default value. Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Directory too difficult?
This is being forwarded to my People who should be banned from using technology folder. - Original Message - From: Colin Anderson To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, January 11, 2007 12:10 AM Subject: RE: [asterisk-users] Directory too difficult? I got a requirement list just now, with my comments inline: (showing it just for a giggle) User requirement: 1) Directory set up by name - If person calling does not know employee's name, how will they access? -Why, using app_telepathy.so of course! User requirement: 2) Directory set by first /or last name?? -Yes. Now decide which one. User requirement: 3) Not all mobile phones have the albphabet on their dialpads, how do they access our directory? -Shout really loud. Telus should have a class action against it for selling Razrs with no DTMF. -Original Message- From: Bryan M. Johns [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 10, 2007 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Directory too difficult? Exactly. ESU = Equipment Superior to Users ;-) Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 10, 2007, at 12:47 PM, Alexander Lopez wrote: More like a ID-10-T error….. -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryan M. Johns Sent: Wednesday, January 10, 2007 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Directory too difficult? I wish had some pearl of wisdom here, but I don't. I will simply share my sympathy. Sounds like an ESU situation to me. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote: I have a group of users whos complaint about Asterisk is that the directory application is too hard too use. (yeah, yeah, I know. For the record, they're Calgarians) Now I'm in a pickle: I don't want to have to create a custom directory for these guys. Anyone have any tips for making the directory easier, maybe re-record the prompts so they are more verbose? We go by first name. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Directory too difficult?
Get me a F*ckin human being seems to work well for me with Verizon. - Original Message - From: Andrew M Stemen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 12, 2007 4:54 AM Subject: Re: [asterisk-users] Directory too difficult? I wouldn't swear to it (chuckle) but each time I've spoken with the Verizon Telecommunications IVR, it sends me to an agent shortly after I start swearing at it. I've had better luck with that than saying things like agent, operator, and attendant. Other than that, it appears to be a rather nice IVR, and it usually understands me. On Thu, 11 Jan 2007 17:10:05 -0500, Paul [EMAIL PROTECTED] said: Or maybe there is a distinct click as the caller prepares to shoot himself in the head. Colin Anderson wrote: If you say: Agent you are transferred to a person. The IVR clearly states that when you call in. I got a demo of Mitel's speech platform last year and it has algorithms that measure apparent stress in a voice. If the voice sounds to stressed, it transfers to an operator. -Original Message- From: Chris Bagnall [mailto:[EMAIL PROTECTED] Sent: Thursday, January 11, 2007 12:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Directory too difficult? the change of Telus' (the ILEC) customer service system entirely to speech recognition. It actually works really, really well I've never been able to screw it up What happens if you yell I just want to talk to a human being! really loudly at it? ;-) Regards, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Andrew Michael Stemen [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording queue calls after an xfer?
I have a problem where my recorded queue calls stop recording once the call is transferred to a different extension. Is there some additional parameter I need to set so recording continues? Is it even possible to do this? Thanks, Jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read Voicmail Boxes
Am Montag, den 15.01.2007, 13:38 -0700 schrieb Andrew Niemantsverdriet:
If you would bother to read my post you will see that what I am
wanting to do is not the asterisk directory cmd. I don't want them to
be able to search or anything fancy like that. I want an app that will
go through and say the recorded name for everyone that has a mailbox
one by one. I did search but was not able to find anything that can do
what I want.
As far as the English class goes; please be more careful about reading
posts and trying to understand exactly what is being asked before you
flame me for not searching.
I have to admit I just scanned over your first post and did
misunderstand it exactly the same way. Hey, there are lots of non-native
speakers around here. No reason for flames though.
I prefer having a productive-to-rant ratio somewhere above 1. So here
comes what I would do:
Write an AGI that walks
through /var/spool/asterisk/voicemail/${YOURCONTEXT}, reads all
directories, and for those directories that contain a file greet.wav,
say the directory name (which most probably is a number, and if you did
reasonably choose voicemail numbers, the phone number is at least
similar to the voicemail number) and playback the greet file.
Here comes what works for me (yeah, I hacked that together just a minute
ago):
/etc/asterisk/extensions.conf:
exten = 411,1,Answer
exten = 411,2,Wait(1)
exten = 411,3,Playback(my-sounds/announcement-for-directory)
exten = 411,4,AGI(vmaildirectory)
exten = 411,5,Playback(my-sounds/now-you-know-the-full-story)
exten = 411,6,Hangup
/usr/share/asterisk/agi-bin/vmaildirectory:
#!/bin/bash
sleep 1
for A in /var/spool/asterisk/voicemail/default/[0-9][0-9][0-9]/greet.wav
do
echo EXEC Playback my-sounds/user
echo EXEC Playback ${A:0:47}
echo EXEC Playback my-sounds/is-at-extension
echo EXEC SayNumber ${A:38:3}
sleep 1
done
(and set the execute flag yadayada).
HTH
Anselm
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Re: [asterisk-users] ANY ADVICE ON THIS????
Thank you, I wasn't aware of the prune command! -Lars On 1/15/07, David Thomas [EMAIL PROTECTED] wrote: On 1/15/07, Lars Knopf [EMAIL PROTECTED] wrote: Hello List, I am stuck with this problem for several days... anybody can give me a hint on this?? I know many of you dealt with problems similar to this, how did you address this?? Thanks in advance!!! -lars -- Forwarded message -- From: Lars Knopf [EMAIL PROTECTED] Date: Jan 11, 2007 1:12 PM Subject: realtime sipusers and rtcachefriends... big headache!! To: asterisk-users@lists.digium.com hi folks, I am using asterisk 1.2.13 (debian etch). My customer's sip accounts are stored in realtime sipusers. I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes Each account has nat=yes Now, I have lot of problems. for example, when I change the 'secret' field of a user in the database, it doesn't get reflected in Asterisk, who is still expecting the old password. Randomly, when trying to dial SIP/x (a customer's account), especially those behind NAT, I get in the console the error no route to Sometimes, too, they can't even register with asterisk. It seems to happen mostly when using realtime. I was digging into the bug tracking system, and I see two bugs that seems to be related, but I can't figure how can I fix it or what step I am supposed to do. The bugs are: http://bugs.digium.com/view.php?id=4687 http://bugs.digium.com/view.php?id=4832 So please, anything than can bring me some light on this... is very appreciated. I think you will need to prune the user/peer after changes. I believe the syntax is something like sip prune realtime user_or_peer where user_or_peer is the actual username. - David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] connecting 2 asterisk servers through OpenVPN
O.Kamal wrote: I am trying to connect 2 asterisk servers through OpenVPN, the VPN should carry 16 channel, however when active channels reached 4 concurrent channels, the connection became unstable, with a very high latency (around 900ms), the internet bandwidth is 1Mbps on each server, I have upgraded the bandwidth to double it, but still have exactly the same problem. Any tips or recommendations on such setup? I am using SIP and G729 between the 2 servers, openVPN using UDP with no compression. Maybe you need more CPU resources for the vpn. In that case you would see problems even if both endpoints were on a gigabit LAN. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording queue calls after an xfer?
1.2 series ? I think that 1.4 has that fixed. At least, that's what my team leaders are telling me ;) Julian. Jay Moore wrote: I have a problem where my recorded queue calls stop recording once the call is transferred to a different extension. Is there some additional parameter I need to set so recording continues? Is it even possible to do this? Thanks, Jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue cmd option 'i'
On 1/15/07, James Fromm [EMAIL PROTECTED] wrote: Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should ignore call forward requests from queue members and do nothing when they are requested. Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding should be ignored. Using Queue(customerservice|i) causes Asterisk to crash when sending the call to the member with forwarding enabled on their SIP device. Am I misinterpreting what this option does? You're not misinterpreting. If it crashes, please file a bug at bugs.digium.com. Thanks. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
Silly question: how are the calls going out? If they're going out through an analog line without the ability to detect hang-ups, then, that's the problem. calls are coming in and out thru an Asterisk server using iax2. have tried two different DID providers and have same problem. Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Directory too difficult?
Just as a followup, I'll intimate to the list what went down: (paraphrased) I'm in Edmonton, the users are in Calgary, so I conferenced in to a Calgary conference with all of the suits in the big boardroom. I basically let them argue themselves in circles about how the Directory app should work in order that they may exhaust themselves, then I politely informed them that the Directory app operated basically the same as on the old Meridian, and that ran for years without comment, so why was it an issue now? And, as it turned out, the one manager who was raising a stink about the directory app actually had never *used* the directory, he had just heard from a friend that Asterisk was crap and we were fools for using it, so he was looking for any excuse to go back to Meridian. I asked for metrics to back up that assertion, he had none, I said: I have some metrics as to how effective Asterisk is and ran phpMyAdmin queries in real time to show total number of calls (a lot), how many buggered up (very, very few), how many minutes we had saved in long distance (a lot), how fast we could provision a new user with a handset and a DID (really really fast), how much the handsets cost (cheaper than Meridian, 1/3 the price of a Mitel) and overall what the platform had cost us (not a lot). He then countered with how the directory was useless because almost no one he knew had letters on his mobile phone and I asked how many people in the boardroom had their mobiles with them, all of them said we all have them, I said Pull them out, and if even one of you has a keypad with no letters on it, I will pull Asterisk out tomorrow and replace it with Meridian. They all did, and predictably all of them had letters on the keypads. A long silence ensued. Then, they thanked me, and apologized for wasting my time. So in the end, the guy who made this stink wound up looking really really stupid in front of his peers, and the suits in Calgary have a new appreciation for how cool Asterisk is. To throw them a bone, I did propose a higher level directory on top of the Directory application so they can press 1 for sales, 2 for accounting etc and told them that they would have to work out amongst themselves what would happen i.e. who would pick up the phone if someone pressed 1 or 2. I haven't heard back from them yet, presumably they are still arguing about it. Lessons learned: 1. If your install is good, compile stats that you can yank up at a moments notice and wave under the nose of any nay-sayer. Stats speak for themselves. 2. Be diplomatic. If I had flippantly told the suits that this was an ID-10-T error, I'd be in Calgary right now installing a Meridian. -Original Message- From: Dovid B [mailto:[EMAIL PROTECTED] Sent: Monday, January 15, 2007 2:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Directory too difficult? This is being forwarded to my People who should be banned from using technology folder. - Original Message - From: Colin Anderson mailto:[EMAIL PROTECTED] To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Sent: Thursday, January 11, 2007 12:10 AM Subject: RE: [asterisk-users] Directory too difficult? I got a requirement list just now, with my comments inline: (showing it just for a giggle) User requirement: 1) Directory set up by name - If person calling does not know employee's name, how will they access? -Why, using app_telepathy.so of course! User requirement: 2) Directory set by first /or last name?? -Yes. Now decide which one. User requirement: 3) Not all mobile phones have the albphabet on their dialpads, how do they access our directory? -Shout really loud. Telus should have a class action against it for selling Razrs with no DTMF. -Original Message- From: Bryan M. Johns [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 10, 2007 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Directory too difficult? Exactly. ESU = Equipment Superior to Users ;-) Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com/ http://www.sheltonjohns.com On Jan 10, 2007, at 12:47 PM, Alexander Lopez wrote: More like a ID-10-T error….. _ From: [EMAIL PROTECTED] [ mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] On Behalf Of Bryan M. Johns Sent: Wednesday, January 10, 2007 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Directory too difficult? I wish had some pearl of wisdom here, but I don't. I will simply share my sympathy. Sounds like an ESU situation to me. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216
Re: [asterisk-users] connecting 2 asterisk servers through OpenVPN
On Mon, 15 Jan 2007, O.Kamal wrote: I am trying to connect 2 asterisk servers through OpenVPN, the VPN should carry 16 channel, however when active channels reached 4 concurrent channels, the connection became unstable, with a very high latency (around 900ms), the internet bandwidth is 1Mbps on each server, I have upgraded the bandwidth to double it, but still have exactly the same problem. Any tips or recommendations on such setup? No real answers, but questions that might help ... Have you tried it without using OpenVPN? Just port-forward the SIP RTP ports, if you need to and give it a go. I am using SIP and G729 between the 2 servers, openVPN using UDP with no compression. Why not IAX? Are your openVPN end-points up to it? Doing high-grade encryption in software might challenge some slower processors - are the VPN endpoints the asterisk boxes themselves? Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording queue calls after an xfer?
Yeah. 1.2.14. I heard bad things about 1.4 not being all that stable. I'm hesitant to move to it. Jay Julian Lyndon-Smith wrote: 1.2 series ? I think that 1.4 has that fixed. At least, that's what my team leaders are telling me ;) Julian. Jay Moore wrote: I have a problem where my recorded queue calls stop recording once the call is transferred to a different extension. Is there some additional parameter I need to set so recording continues? Is it even possible to do this? Thanks, Jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] S400M (FXS) Modules no longer seen
Hi, I have a TDM21B (A new TDM20B that just arrived plus a X400M that I already had). This card sometimes works, and sometimes only the FXO module is seen. By works, I mean all 3 modules have the green lights on the ports on and when zaptel is loaded the log shows: Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) When things don't work, I either see: Indirect Registers failed verification. [deleted registers in question] Module 0: FAILED FXS (FCC) Indirect Registers failed verification. [deleted registers in question] Module 1: FAILED FXS (FCC) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Or sometimes, I'd get nothing about FXS modules, and only log messages about the FXO working. I removed the FXO so that I have just what was just shipped to me. Now nothing is recognized. If I manually insmod zaptel, all I see is: Found a Wildcard TDM: Wildcard TDM400P REV I Any ideas? Before sending this card back, I'd like to make sure there isn't something else basic that I'm not doing. I did supply power to the TDM400P. That was the first thing I checked. Thanks, MikeC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Queue cmd option 'i'
-Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Monday, January 15, 2007 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue cmd option 'i' On 1/15/07, James Fromm [EMAIL PROTECTED] wrote: Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should ignore call forward requests from queue members and do nothing when they are requested. Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding should be ignored. Using Queue(customerservice|i) causes Asterisk to crash when sending the call to the member with forwarding enabled on their SIP device. Am I misinterpreting what this option does? You're not misinterpreting. If it crashes, please file a bug at bugs.digium.com. Thanks. I wonder how this could actually work? If Asterisk sends an INVITE to a phone, and the phone responds with 'Moved Temporarily', and Asterisk sends the INVITE again, isn't the phone just going to send 'Moved Temporarily' again? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
Am Montag, den 15.01.2007, 14:22 -0800 schrieb chester c young: Silly question: how are the calls going out? If they're going out through an analog line without the ability to detect hang-ups, then, that's the problem. calls are coming in and out thru an Asterisk server using iax2. have tried two different DID providers and have same problem. Chester, could you verify or negate that adding the T option makes it work? Did you look if there is a bug report somewhere that has to do with call teardown problems when Asterisk is not in the Audio path? BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
Anselm Martin Hoffmeister wrote: Am Montag, den 15.01.2007, 14:22 -0800 schrieb chester c young: Silly question: how are the calls going out? If they're going out through an analog line without the ability to detect hang-ups, then, that's the problem. calls are coming in and out thru an Asterisk server using iax2. have tried two different DID providers and have same problem. Chester, could you verify or negate that adding the T option makes it work? Did you look if there is a bug report somewhere that has to do with call teardown problems when Asterisk is not in the Audio path? Curious - is this still a $50 thread? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1-way audio
I know when you read that subject everyone thinks NAT, but that isn't the case here. Incoming calls get 2 way audio, but outbound calls do not have incoming audio. below is the flow callee -- asterisk -- firewall/router -- provider Callee is firewalled, but not NAT. callee is on the same subnet as the asterisk box. Asterisk box has been completely excluded from the firewall rules, and all traffic is being passed to it. Provider says everything is good on their end. Incoming calls work fine. Here is the config from sip.conf for the provider: [general] context=incoming srvlookup=yes canreinvite=no videosupport=yes qualify=yes ; trunk links [provider] type=friend nat=never host=123.123.123.123 disallow=all allow=ulaw dtmfmode=auto context=dids-inbound canreinvite=no Thanks Miles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] S110M (FXS) Modules no longer seen on TDM400P
I have a TDM21B (A new TDM20B that just arrived plus a X100M that I already had). This card sometimes works, and sometimes only the FXO module is seen. (correction to previous message that incorrectly listed X400M/S400M when I'm using only the single port versions of these modules (X100M/S110M). By works, I mean all 3 modules have the green lights on the ports on and when zaptel is loaded the log shows: Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) When things don't work, I either see: Indirect Registers failed verification. [deleted registers in question] Module 0: FAILED FXS (FCC) Indirect Registers failed verification. [deleted registers in question] Module 1: FAILED FXS (FCC) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Or sometimes, I'd get nothing about FXS modules, and only log messages about the FXO working. I removed the FXO so that I have just what was just shipped to me. Now nothing is recognized. If I manually insmod zaptel, all I see is: Found a Wildcard TDM: Wildcard TDM400P REV I Any ideas? Before sending this card back, I'd like to make sure there isn't something else basic that I'm not doing. I did supply power to the TDM400P. That was the first thing I checked. Thanks, MikeC -- Michael C. Cambria email : [EMAIL PROTECTED] VoIP : sip:[EMAIL PROTECTED] FWD : sip:[EMAIL PROTECTED] WWW : www.fid4.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
--- Paul [EMAIL PROTECTED] wrote: Anselm Martin Hoffmeister wrote: Curious - is this still a $50 thread? yes. Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Practical limit on dial prefixes for a route
Colleagues, We're in the process of standardizing on Sprint PCS and Cingular phones on a national basis (~ 50 properties, 1000's of lines). I manage an Asterisk install at one location. I've been looking at the Multitech CellFinder CDMA for Sprint as a dial backup solution. Basically, it's a CDMA to POTS gateway, tied to a PCS account. We would see it as a trunk line and I would like to do LCR and route out the CellFinder line(s)^ all PCS calls, since we have free PCS to PCS. Here's the kicker. Since we're on a natioinal basis, it would make sense to have a large LCR listing of prefixes reachable from the gateway, which would most likely number in the thousands of prefixes. Has anyone encountered an upper practical limit that * has for prefixes reachable via a route. I assume that search time is somewhat of a factor. The * box doing the routing is a dual core machine with 4GB of RAM, so it has lots of horsepower. Wondering what limits users have pushed it to on a large scale. Could it handle something like that or would it implode from a huge routing table (assuming our tech contacts at PCS could supply us with a national listing of NPA-NXX's on the PCS network). Thanks in advance for any info. EKG ^ depending on call volume, we may install multiple cell lines ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Practical limit on dial prefixes for a route
Correction, that's Multitech CALLFinder CDMA, not CellFinder. Sorry for the misquote. EKG -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Germann Sent: Monday, January 15, 2007 8:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Practical limit on dial prefixes for a route Colleagues, We're in the process of standardizing on Sprint PCS and Cingular phones on a national basis (~ 50 properties, 1000's of lines). I manage an Asterisk install at one location. I've been looking at the Multitech CellFinder CDMA for Sprint as a dial backup solution. Basically, it's a CDMA to POTS gateway, tied to a PCS account. We would see it as a trunk line and I would like to do LCR and route out the CellFinder line(s)^ all PCS calls, since we have free PCS to PCS. Here's the kicker. Since we're on a natioinal basis, it would make sense to have a large LCR listing of prefixes reachable from the gateway, which would most likely number in the thousands of prefixes. Has anyone encountered an upper practical limit that * has for prefixes reachable via a route. I assume that search time is somewhat of a factor. The * box doing the routing is a dual core machine with 4GB of RAM, so it has lots of horsepower. Wondering what limits users have pushed it to on a large scale. Could it handle something like that or would it implode from a huge routing table (assuming our tech contacts at PCS could supply us with a national listing of NPA-NXX's on the PCS network). Thanks in advance for any info. EKG ^ depending on call volume, we may install multiple cell lines ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] php agi - first phrase truncated, all others fine
Thanks all, I'll give this a shot.
On 1/15/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: *blackwater dev [EMAIL PROTECTED]
Yes, the wav file is fine. For some reason it's just getting cut off.
Whatever I type there seems to get cut off, strange.
With plain dialplan (no AGI), I notice that the first few syllables from
Playback() or Background() could be eaten up, sometimes more, sometimes
less. A little Wait() after Answer() may help stablize the voice channel.
Yuan Liu
On 1/15/07, Paul [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Are you creating a temporary wav file? If so, look at that first. If the
wav file is truncated at least you know the problem is related to the
way swift gets invoked and passed the text argument. If the file is okay
you need to look at the way it gets handled afterwards.
blackwater dev wrote:
I have the following code. When I call the extension, it either
ignores the first Hello there everyone, or says hello and moves on
sometime stoping before it finishes hello. The rest of the text reads
fine. Anyone else have this issue??
Thanks!
require('/var/lib/asterisk/agi-bin/phpagi.php');
$agi = new AGI();
$agi-answer();
$agi-swift(Hello there everyone );
$agi-swift(Please press 1 for a search .);
$result= $agi-get_data('beep',3, 1);
$zip= $result['result'];
$agi-swift(That concludes your call. Thank you, Good bye .);
$agi-hangup();
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RE: [asterisk-users] Practical limit on dial prefixes for a route
Why not use DBGet / DBPut? I use it for Caller ID and I have over 50K entries in the DB, and there is no appreciable load hitting the DB in the dialplan. And my one install (admittedly modest) hits the DB a few thousand times a day, with up to 46 concurrent calls. -Original Message- From: Eric Germann To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: 1/15/2007 6:34 PM Subject: RE: [asterisk-users] Practical limit on dial prefixes for a route Correction, that's Multitech CALLFinder CDMA, not CellFinder. Sorry for the misquote. EKG -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Germann Sent: Monday, January 15, 2007 8:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Practical limit on dial prefixes for a route Colleagues, We're in the process of standardizing on Sprint PCS and Cingular phones on a national basis (~ 50 properties, 1000's of lines). I manage an Asterisk install at one location. I've been looking at the Multitech CellFinder CDMA for Sprint as a dial backup solution. Basically, it's a CDMA to POTS gateway, tied to a PCS account. We would see it as a trunk line and I would like to do LCR and route out the CellFinder line(s)^ all PCS calls, since we have free PCS to PCS. Here's the kicker. Since we're on a natioinal basis, it would make sense to have a large LCR listing of prefixes reachable from the gateway, which would most likely number in the thousands of prefixes. Has anyone encountered an upper practical limit that * has for prefixes reachable via a route. I assume that search time is somewhat of a factor. The * box doing the routing is a dual core machine with 4GB of RAM, so it has lots of horsepower. Wondering what limits users have pushed it to on a large scale. Could it handle something like that or would it implode from a huge routing table (assuming our tech contacts at PCS could supply us with a national listing of NPA-NXX's on the PCS network). Thanks in advance for any info. EKG ^ depending on call volume, we may install multiple cell lines ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Practical limit on dial prefixes for a route
Eric Germann wrote: Colleagues, We're in the process of standardizing on Sprint PCS and Cingular phones on a national basis (~ 50 properties, 1000's of lines). I manage an Asterisk install at one location. I've been looking at the Multitech CellFinder CDMA for Sprint as a dial backup solution. Basically, it's a CDMA to POTS gateway, tied to a PCS account. We would see it as a trunk line and I would like to do LCR and route out the CellFinder line(s)^ all PCS calls, since we have free PCS to PCS. Two comments: Cingular is GSM, Sprint is CDMA With LNP , NPA-NXX isn't enough information to determine free on network calling Since wireline to wireless LNP, the NPA assignments are no longer locked to a specific carrier. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
