Re: [asterisk-users] Route an incoming call by ANI*DNIS
Dan Casey wrote:
Sorry for my very delayed response. To answer a few questions:
1. Right, the *ANI*DNIS* is not working correctly. When the telco sends
it, we are always missing the beginning of it. I almost always get a 7
digit ani, but sometimes it is 8 or 6.
I won't be able to help with hardware part, but there's a simple trick
to get them as you want:
[incoming]
_X.,1,Set(DNIS=${CUT(${EXTEN:-4})})
_X.,2,Goto,dnis,${DNIS},1
[dnis]
6789 = ...
So, passing 123456789 or 23456789 or 0123456789 would all go to context
dnis, extension 6789.
Regards,
Atis
2. The phone company assured me several times that we are not set up for
feature group d. I tried anyway and it had the same affect. Got a
non-Feature Group D input on channel 77. Assuming EM Wink instead..
Pretty smart.
3. I've also triad featb featdmf and featdmf_ta. All of which were the
same or much works. Featb atcually crashed my polycom phone when I
tried to dial into the pbx. :)
4. I didn't try 5ess as were are not using a pri. After this whole
ordeal, where contemplating having all our t1's switched to pri. That
would really be a bad thing, but its a pretty drastic measure.
5. The callerid issue is partially solved. I handled it with
CALLERID=$DID. The main phone system runs a perl script to parse, and
handle the rest of it. Which actually I like this better, because I
don't have to setup extensions for every DNIS.. I have about 50 of them.
We are still having the same signaling issue with both XO and GBX.
Heres the problem I'm getting in asterisk.
1. A wink issue
Nov 19 05:35:06 DEBUG[12565] chan_zap.c: Got wink in weird state 4 on
channel 92
2. Calls that drop randomly
Nov 19 12:29:51 DEBUG[15370] channel.c: Avoiding initial deadlock for
'SIP/1026-b7901568'
Nov 19 12:31:02 DEBUG[22546] channel.c: Didn't get a frame from channel:
SIP/1026-b7901568
Nov 19 12:31:02 DEBUG[22546] channel.c: Bridge stops bridging channels
SIP/1026-b7901568 and Zap/1-1
3. Two phone companies that say I'm sending too many winks went the
seizure, and a Digium rep who says it should just work.
I am shot...
Surely I can't be the only person who has this problem. Is there anyone
else not using PRI, who had issues like this.??
Dan
Jon Weisman wrote:
Dan,
What happen w/ this? Did you figure it out? I've setup w/ XO using ESF/B8ZS,
and 5ESS for the switchtype, worked great and got the ANI as well. I dont
think you can get ANI on EM Wink trunks, how about feature group d?
-Jon
- Original Message -
From: Dan Casey [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, November 02, 2007 9:47 AM
Subject: [asterisk-users] Route an incoming call by ANI*DNIS
does anyone know how to route a call coming in with ANI*DNIS*
Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing
Set(Zap/49-1, DID=1231234*4812*) in new stack
I tried making a route for _.*4812* but that matched everything rather
then just the dnis i wanted.. any ideas?
I would preferably like pass the callerid along to my extensions, but
for now the important thing is routing.
Thanks
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[asterisk-users] Problems with losing D-Channel on
Hello all, I got a problem at an asterisk server, with dropping calls, losing all channels and reaktivating all channels and beeing back up. This problem seems to occure randomly over the whole day, when it gots traffic on the card. After looking @ google I found several hints but none did work fine. To avoid problems with the phone line (german E1) I called the provider, he did a 45 min. route test with incoming and outgoing calls over all lines without any problem over the whole time. I also got a phone call with the providers service partner for the S2M part. He reset the line and putt he errorcounter to 0. After the test it was still on 0. When we plugged in the card again, there were again errors on the counter after ~ 5-10 minutes. After this, i put the asterisk, zaptel and libpri versions to newest versions, now it's working a bit better, but after 1 day fine work, it crashes again all calls. The system worked fine about 6 month now, but since 2 weeks I got the problems. Does anyone have any idea? The last points what is on my todo is to switch the pci slot. The cable which connects card to E1 interface is also switched. Kind regards, Eric I got a snippet from the console, where the problem is occuring. [Nov 16 15:57:09] ERROR[5499]: chan_zap.c:8178 zt_pri_error: Write to 39 failed: Unknown error 500 [Nov 16 15:57:09] ERROR[5499]: chan_zap.c:8178 zt_pri_error: Short write: 0/15 (Unknown error 500) [Nov 16 15:57:09] WARNING[5499]: chan_zap.c:3822 zt_handle_event: Detected alarm on channel 1: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 2: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 2 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 6: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 6 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 7: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 7 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 8: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 8 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 9: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 9 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 10: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 10 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 11: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 11 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 12: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 12 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 13: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 13 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 14: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 14 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 15: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 15 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 17: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 17 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 18: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 18 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 19: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 19 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 20: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 20 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 21: Red Alarm [Nov 16 15:57:09]
[asterisk-users] sl75 wlan not able of being pickuped?
Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log of such a call. Any help would be high appreciated. regards t. asterix*CLI sip debug SIP Debugging enabled asterix*CLI -- SIP read from 217.10.79.9:5060: --- (0 headers 0 lines) Nat keepalive --- asterix*CLI -- SIP read from 192.168.150.51:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 70 Content-Length: 293 Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK6abcc4ace Call-ID: 94cba353ee1163b From: Steffen sip:[EMAIL PROTECTED];tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED] CSeq: 2078851383 INVITE Supported: timer Session-Expires: 7200 Allow-Events: talk, hold, conference Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO Content-Type: application/sdp Contact: Steffen sip:[EMAIL PROTECTED]:5060;transport=udp Supported: replaces User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 v=0 o=MxSIP 0 1730916047 IN IP4 192.168.150.51 s=SIP Call c=IN IP4 192.168.150.51 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (16 headers 13 lines) --- Using INVITE request as basis request - 94cba353ee1163b Sending to 192.168.150.51 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.150.51:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK6abcc4ace;received=192.168.150.51 From: Steffen sip:[EMAIL PROTECTED];tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED];tag=as06838deb Call-ID: 94cba353ee1163b CSeq: 2078851383 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7e84319d Content-Length: 0 --- Scheduling destruction of call '94cba353ee1163b' in 15000 ms Found user '116' asterix*CLI -- SIP read from 192.168.150.51:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK6abcc4ace Call-ID: 94cba353ee1163b From: Steffen sip:[EMAIL PROTECTED];tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED];tag=as06838deb CSeq: 2078851383 ACK User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 --- (9 headers 0 lines) --- asterix*CLI -- SIP read from 192.168.150.51:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 70 Content-Length: 293 Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK2e49558fa Call-ID: 94cba353ee1163b From: Steffen sip:[EMAIL PROTECTED];tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED] CSeq: 2078851384 INVITE Supported: timer Session-Expires: 7200 Allow-Events: talk, hold, conference Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO Content-Type: application/sdp Proxy-Authorization:Digest response=ea33e742f1b16d49344c67d8cc980a16,username=116,realm=asterisk,nonce=7e84319d,algorithm=MD5,uri=sip:[EMAIL PROTECTED] Supported: replaces Contact: Steffen sip:[EMAIL PROTECTED]:5060;transport=udp User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 v=0 o=MxSIP 0 1730916047 IN IP4 192.168.150.51 s=SIP Call c=IN IP4 192.168.150.51 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (17 headers 13 lines) --- Using INVITE request as basis request - 94cba353ee1163b Sending to 192.168.150.51 : 5060 (non-NAT) Found user '116' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 9 Found RTP audio format 101 Peer audio RTP is at port 192.168.150.51:5004 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G722 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw| g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 119 in default (domain 192.168.150.151) list_route: hop: sip:[EMAIL PROTECTED]:5060;transport=udp Transmitting (no NAT) to 192.168.150.51:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK2e49558fa;received=192.168.150.51 From: Steffen sip:[EMAIL PROTECTED];tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED] Call-ID: 94cba353ee1163b CSeq: 2078851384 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- We're at
Re: [asterisk-users] California based PSTN connections
Just a follow up, I have my server with Cari.net in San Diego. How do you go about getting a block of DIDs and performing my own origination? Anyone has any experience in this field? Thanks. On 11/19/07, Eric Chamberlain [EMAIL PROTECTED] wrote: We use VoicePulse Connect. They now have a POP in San Francisco. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Adrian Marsh *Sent:* Saturday, November 17, 2007 5:33 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] California based PSTN connections Hi, Can anyone recommend any company that can provide PSTN termination for SIP calls, at least USA based, preferably California area. One of my A*k servers is US based and I don't want my traffic flowing back and forth via my current UK PSTN provider for USUS calls. Thanks, Adrian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime - mysql query gives wrong results??
Hi, I am using Realtime for sip configuration. When there is an INVITE which arrives at asterisk asterisk makes the following selects: Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE name = 'tzl' [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE host = '192.168.0.74' AND port = '5060' [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE ipaddr = '192.168.0.74' AND port = '5060' [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:258 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE host = '192.168.0.74' ORDER BY host [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:258 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE ipaddr = '192.168.0.74' ORDER BY ipaddr Found no matching peer or user for '192.168.0.74:5060' as seen above there is no result for this select although I have such record in the database: mysql SELECT * FROM sip_conf WHERE host = '192.168.0.74' ORDER BY host; ++---+-+--+---++-+-+---+--+--++-+--+--+--+-+---+-+--++--+-+--+-+-++++--+--+--+-+-+++--+++ | id | name | accountcode | amaflags | callgroup | callerid | canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | fullcontact | host | insecure | language | mailbox | md5secret | nat | deny | permit | mask | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | type | username | disallow | allow | musiconhold | regseconds | ipaddr | regexten | cancallforward | setvar | ++---+-+--+---++-+-+---+--+--++-+--+--+--+-+---+-+--++--+-+--+-+-++++--+--+--+-+-+++--+++ | 3 | outbound-voip.touk.pl | NULL| NULL | NULL | TouK S.K.A | no | NULL| NULL | NULL | NULL | NULL | NULL| 192.168.0.74 | NULL | NULL | NULL | NULL | no | NULL | NULL | NULL | NULL| | NULL | NULL| NULL | NULL | NULL | peer | | all | g729;ilbc;gsm;ulaw;alaw | NULL| 0 | | | yes|| ++---+-+--+---++-+-+---+--+--++-+--+--+--+-+---+-+--++--+-+--+-+-++++--+--+--+-+-+++--+++ Why does realtime select give no results?? Cheers tomasz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channels to destroy
Hello, On Nov/19/2007, Johansson Olle E wrote: 16 nov 2007 kl. 14.06 skrev Carles Pina i Estany: In a couple of Asterisks, after type sip show channels we have a lot of these: IP_PEER dst_number something00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE [...] Is it normal? How we can remove it? Depending on the traffic on your server and whether they disappear finally after a while or hang forever, it may be a bug. Please try with the latest 1.2 version, since we spent a lot of time fixing these kind of issues earlier this year. Or even better, take time to update to version 1.4, since 1.2 is not maintained any more. ok, I will try but it will take some time. Thanks for your answer :-) Anyway, one more question: this BYE channels that doesn't disappear, can cause any problem? If the problem still persists in 1.4, please file a bug report and we'll start working on it. I will -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MediaHandling--Help Required
Hello Users, My Setup is like this openser --Registrar asterisk --Callflow using asterisk-b2bua + radius for accounting My Intention was to generate a Acct-Stop Packet when there is a failure of RTP media from one of the UAC's( callee or caller) who is in dialog. so that the Caller will not be charged for Meaning less network problems Because there is no way asterisk knows about failed UAC as he may not send a BYE Packet . i used the following parameters set canreinvite=no; rtptimeout=60 seconds; Still there is no Acct-Stop packet generated until the session expires timer fires which is equal to Session-Timeout value from radius? Can anybody have any idea of handling network problem of his type? Looking forward for suggestions Thanks in advance srinivas antarvedi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SMS Feature In Asterisk
Is Asterisk capable of sending text messages to a cell phone or is there an application for that? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS Feature In Asterisk
broadband Voice wrote: Is Asterisk capable of sending text messages to a cell phone or is there an application for that? Yes. Any carrier that supports SMS over analog lines will work with the Asterisk SMS application. Generally carriers in the USA and Canada do not support SMS over analog lines, but do generally have an e-mail-SMS gateway. Check with your carrier. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] not sending bye
Hello, We are using this Asterisk: 1.2.14-BRIstuffed-0.3.0-PRE-1y Everything works fine but we have an issue (not often, but one call every some hundreds) I sniffed the communication between phone, Asterisk and softswitch. I can see that Asterisk receives a Cancel from phone but Asterisk never sends a Cancel to Softswitch. This makes us some problems: billing system doesn't allow next call because there is a call limit (1 per extension), etc. Why Asterisk receives Cancel and never sends Cancel? But this happends only sometimes, not always. Yes, as soon as I get the chance I will update this Asterisk. But somebody could tell me why this is happening? I browsed in internet to find some bugreport with same behaviour without any luck. I would like to find some bug report with same problem and fine that is fixed for next Asterisk version :-) Thanks, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS Feature In Asterisk
On Nov 20, 2007 6:24 AM, Eric ManxPower Wieling wrote: [...] but do generally have an e-mail-SMS gateway. Check with your carrier. http://en.wikipedia.org/wiki/SMS_gateways -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] store 2 separate records in cdr when a call is transferd
Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] store 2 separate records in cdr when a call is transferd
nik600 wrote: Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all You want to do that on blind transfer or attended transfer? I got it working on blindxfer - it's pretty simple. Do a ResetCDR(w) in the context defined within TRANSFER_CONTEXT var. Attended transfers are much more nightmare for CDRs.. There are several channels involved, so it would need some cleaning to get what you want (i just don't use them) Regards, Atis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] r2 multiframe error - solved
All errors was genereted by physical link. Protocolvariant cz,10,6 its ok for me in Poland Thanks for help Regards Akron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to enable res_config_mysql
okay, probably a typing issue check in extconfig.conf you have a line that is something like sipusers = mysql.asterisk_4,some_table_blablabla and it should be sipusers = mysql,asterisk_4,some_table_blablabla Note the change from period to comma right after mysql. Otherwise post that section of your extconfig.conf Make sure you have the table created. Hi, I have a new issue now:) I compiled module and put it to modules dir configured the res_mysql.conf file but when asterisk tries to take some data from db I get: [Nov 19 17:10:00] WARNING[2801]: config.c:1235 find_engine: Realtime mapping for 'sipusers' found to engine 'mysql.asterisk1_4', but the engine is not available ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime - mysql query gives wrong results??
Two things: 1. Set the context 2. Set the port Hi, I am using Realtime for sip configuration. When there is an INVITE which arrives at asterisk asterisk makes the following selects: Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE name = 'tzl' [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE host = '192.168.0.74' AND port = '5060' [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE ipaddr = '192.168.0.74' AND port = '5060' [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:258 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE host = '192.168.0.74' ORDER BY host [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:258 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE ipaddr = '192.168.0.74' ORDER BY ipaddr Found no matching peer or user for '192.168.0.74:5060' as seen above there is no result for this select although I have such record in the database: mysql SELECT * FROM sip_conf WHERE host = '192.168.0.74' ORDER BY host; ++---+-+--+--- ++-+-+---+--+--- ---++-+--+--+--- ---+-+---+-+--++--+- +--+-+-+++-- --+--+--+--+-+-- ---+++--+++ | id | name | accountcode | amaflags | callgroup | callerid | canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | fullcontact | host | insecure | language | mailbox | md5secret | nat | deny | permit | mask | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | type | username | disallow | allow | musiconhold | regseconds | ipaddr | regexten | cancallforward | setvar | ++---+ -+--+---++-+-+-- -+--+--++-+- -+--+--+-+---+-+--+ +--+-+--+-+-++-- --++--+--+--+--- --+-+++--+-- --++ | 3 | outbound-voip.touk.pl | NULL| NULL | NULL | TouK S.K.A | no | NULL| NULL | NULL | NULL | NULL | NULL| 192.168.0.74 | NULL | NULL | NULL | NULL | no | NULL | NULL | NULL | NULL| | NULL | NULL| NULL | NULL | NULL | peer | | all | g729;ilbc;gsm;ulaw;alaw | NULL| 0 | | | yes|| ++--- +-+--+---++-+--- --+---+--+--++- +--+--+--+-+---+-+-- ++--+-+--+-+-+-- --+++--+--+--+-- ---+-+++ --+++ Why does realtime select give no results?? Cheers tomasz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] store 2 separate records in cdr when a call is transferd
for blind transfer! Many thanks! On Nov 20, 2007 2:24 PM, Atis Lezdins [EMAIL PROTECTED] wrote: nik600 wrote: Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all You want to do that on blind transfer or attended transfer? I got it working on blindxfer - it's pretty simple. Do a ResetCDR(w) in the context defined within TRANSFER_CONTEXT var. Attended transfers are much more nightmare for CDRs.. There are several channels involved, so it would need some cleaning to get what you want (i just don't use them) Regards, Atis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - What is Alarm receiver feature ?
Hello, From http://www.asterisk.org/support/features or http://www.voip-info.org/wiki/index.php?page=Asterisk%20Features , there is a features list I'm trying to translate and explain to prospective customers. I can't relate this Alarm receiver feature to anything meaningful. Does it mean anything precise to someone ? Does it mean you can connect an alarm appliance to an Asterisk analog interface ? Does it relate to Asterisk being able to exchange data with Linux system which in turn have many types of interfaces ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - What is Alarm receiver feature ?
Quoting Olivier [EMAIL PROTECTED]: if you are going to be a security company that receives alarm notification from burglar/fire alarms, this is the module for you - otherwise ignore it. Hello, From http://www.asterisk.org/support/features or http://www.voip-info.org/wiki/index.php?page=Asterisk%20Features , there is a features list I'm trying to translate and explain to prospective customers. I can't relate this Alarm receiver feature to anything meaningful. Does it mean anything precise to someone ? Does it mean you can connect an alarm appliance to an Asterisk analog interface ? Does it relate to Asterisk being able to exchange data with Linux system which in turn have many types of interfaces ? Regards Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pass CallerID when call forwards to PSTN?
On Nov 20, 2007 5:06 AM, Dovid B [EMAIL PROTECTED] wrote:
From what I have seen in the past asterisk should pass along the CID
automatically. As some one else already mentioned. It can be your ITSP. You
can always set the CID with Set(CALLERID(num)=1234567890).
Asterisk does pass the caller ID for the internal calls, but for the
external ones, my default outbound CallerID gets used.
I can set a different CID like you suggest above, but I don't know how
to get the inbound CID so I can set it correctly. Does anyone know if
there's a variable exposed to my extensions.conf so I can do something
like Set(CALLERID(num)=${VAR}) and set outbound callerID to that of
the calling party?
Thanks,
Russell
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[asterisk-users] Realtime extensions configuration - calling user filtering
Hi, Is it possible to filter the calling user with the usage of mysql realtime the same as it is done in extensions.conf file: exten = some_exten/calling-user is there some flag which activates this extra check?? Cheers Tomasz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ACD functionality , Skills for agents
Hi all, I have a question regarding ACD for queues. What happens when I have 2 or more queues with same weight and each queue has a call in? How will it decide which call will be routed to the next available agent? Will it take the call with the longest waiting time in queue? If not how would I do this? Also can someone point me to resources for making a single queue with customer calls tagged with agent skills? What I mean is instead of having multiple queues Sales,Tech support, etc, have only a single queue with calls being tagged according to the customer's choice from IVR, so if a customer would choose SALES , the call would go into the queue with other calls but it would only be answered from agents with the skill SALES. This is something offered in other PBX systems like Avaya but im pretty sure it can be done on Asterisk, right? Thanks, Kyriakos Mavromichalis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ACD functionality , Skills for agents
On Nov 20, 2007 10:16 AM, Kyriakos [EMAIL PROTECTED] wrote: I have a question regarding ACD for queues. What happens when I have 2 or more queues with same weight and each queue has a call in? How will it decide which call will be routed to the next available agent? Will it take the call with the longest waiting time in queue? If not how would I do this? Beware of queue weights. They have caused major problems in the past for many people on this list. As I understand it, enabling weights requires * to grab a lock on a large number of data structures related to queue state, which can cause performance slowdowns and crashes. I haven't seen reports of this recently, so it might be better in the later 1.4 releases, but at one time it was a sure-fire recipe for pain. Also can someone point me to resources for making a single queue with customer calls tagged with agent skills? What I mean is instead of having multiple queues Sales,Tech support, etc, have only a single queue with calls being tagged according to the customer's choice from IVR, so if a customer would choose SALES , the call would go into the queue with other calls but it would only be answered from agents with the skill SALES. This is something offered in other PBX systems like Avaya but im pretty sure it can be done on Asterisk, right? It probably could be, but it would make reporting pretty difficult, as the key fields in the queue log are the call id and the queue name. While you could use the QueueLog() application to stick extra data about the call (e.g the skill chosen from the IVR) into the queue log, that would appear in one line only and require post-processing to glue it together with the rest of the data for that call. I'm pretty sure it wouldn't mesh nicely with the reporting package I use (QueueMetrics). What I do for this is maintain queue (skill) membership in a database, then add the channels to the appropriate queues when the agents log on via a web page. Is there a particular reason you want to just have one queue? -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime extensions configuration - calling user filtering
As much I as can tell, Asterisk version 1.2 doesn't support the ex-girlfriend logic that you ask. I didn't test that feature with 1.4 releases, maybe they already implement it. Regards, Ricardo Carvalho.. On Nov 20, 2007 2:51 PM, Tomasz Zieleniewski [EMAIL PROTECTED] wrote: Hi, Is it possible to filter the calling user with the usage of mysql realtime the same as it is done in extensions.conf file: exten = some_exten/calling-user is there some flag which activates this extra check?? Cheers Tomasz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Route an incoming call by ANI*DNIS
On Tuesday 20 November 2007 02:38:38 Atis Lezdins wrote:
I won't be able to help with hardware part, but there's a simple trick
to get them as you want:
[incoming]
_X.,1,Set(DNIS=${CUT(${EXTEN:-4})})
_X.,2,Goto,dnis,${DNIS},1
[dnis]
6789 = ...
I don't think you've actually tested this, because if you had, you would find
that it does not work.
[incoming]
exten = _X.,1,Goto(dnis,${EXTEN:-4},1)
[dnis]
exten = 6789,1,.
--
Tilghman
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Re: [asterisk-users] How to integrate Asterisk with Avaya
which Avaya system? and what are you trying to add with asterisk? On 11/20/07, Dovid B [EMAIL PROTECTED] wrote: Hello Everyone, Can someone please point to sources how to integrate Asterisk PBX with Avaya..? What normalize and expose protocol/API does Avaya support which can be use with Asterisk? Thanks in advance, -C What are you trying to do between asterisk and Avaya ? Avaya has software (forgot the name of it) that allows you to connect to it and set the configs etc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] r2 multiframe error - solved
Jakub Syrek wrote: All errors was genereted by physical link. Protocolvariant cz,10,6 its ok for me in Poland Thanks for help Regards Akron Thanks. I will make a note of that in the code. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pass CallerID when call forwards to PSTN?
On Tuesday 20 November 2007 08:50:06 Russell Horn wrote:
On Nov 20, 2007 5:06 AM, Dovid B [EMAIL PROTECTED] wrote:
From what I have seen in the past asterisk should pass along the CID
automatically. As some one else already mentioned. It can be your ITSP.
You can always set the CID with Set(CALLERID(num)=1234567890).
Asterisk does pass the caller ID for the internal calls, but for the
external ones, my default outbound CallerID gets used.
I can set a different CID like you suggest above, but I don't know how
to get the inbound CID so I can set it correctly. Does anyone know if
there's a variable exposed to my extensions.conf so I can do something
like Set(CALLERID(num)=${VAR}) and set outbound callerID to that of
the calling party?
Callerid is automatically passed along the route of the call. It is
inherited, unless overwritten. If you are using analog lines, this is not
possible. If you are using an ITSP or a PRI, you need to ensure that your
provider will let you set arbitrary CallerID (explain to them the legitimate
purpose of forwarding calls out to cell phones, and they are much more
likely to permit that, than if you say you want to spoof CallerID).
--
Tilghman
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Re: [asterisk-users] r2 multiframe error - solved
Good news. On Nov 20, 2007 7:51 AM, Jakub Syrek [EMAIL PROTECTED] wrote: All errors was genereted by physical link. Protocolvariant cz,10,6 its ok for me in Poland Thanks for help Regards Akron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to integrate Asterisk with Avaya
Hi, Im not talking about connecting Asterisk with Avaya system. I just mentioned Avaya because on a presentation I've been to, they said that this could be done. I want to do this on Asterisk. I already have a call centre setup with 5 different queues with same weights but it seems that ACD is not taking into consideration how long the call has been waiting in each queue and its randomly choosing queue for next available agent. The result is that some calls might be answered even though they have much shorter waiting time in queue than calls to other queues. I would like to create only one queue and have customers choose using an IVR a call category (Sales, Tech support, accounting, etc.), tag the call with a code or something so asterisk will know what this call is about, send it into queue along with other category calls, and when its time to be sent to the next available agent, it should be sent to an agent marked with the same code that the call is tagged with ie if call is tagged for sales then the available agent with the highest weight for skill SALES should answer the calls. This way I make sure I have FIFO for incoming calls. BR KM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, November 20, 2007 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to integrate Asterisk with Avaya which Avaya system? and what are you trying to add with asterisk? On 11/20/07, Dovid B [EMAIL PROTECTED] wrote: Hello Everyone, Can someone please point to sources how to integrate Asterisk PBX with Avaya..? What normalize and expose protocol/API does Avaya support which can be use with Asterisk? Thanks in advance, -C What are you trying to do between asterisk and Avaya ? Avaya has software (forgot the name of it) that allows you to connect to it and set the configs etc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] blind transfer dumping calls
On Mon, 2007-11-19 at 16:26 +0200, Atis Lezdins wrote: On 11/19/07, Brian J. Murrell [EMAIL PROTECTED] wrote: I am using asterisk 1.4.10 and seem to be having a problem with blind transfer. This could very well be a pebkac problem but I'm not sure. This is probably issue with 1.4.10. I have reported it, and it has been fixed in 1.4.10.1. Please see http://bugs.digium.com/view.php?id=10415 Nope. That does not appear to be my problem. I patched the relevant patch in and I still get: -- Started music on hold, class 'default', on channel 'Zap/1-1' -- SIP/1011002206-081fbac0 Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on Zap/1-1 -- Transferring Zap/1-1 to '2005' (context internal-sip) priority 1 == Channel 'Zap/1-1' jumping out of macro 'dialhouse' -- Hungup 'Zap/1-1' When I try to transfer to 2005. 2005 never rings. Given the patch, I have the spot in the code where the transfer is supposed to happen so I could do some debugging I guess. ~sigh~ But this has to be working for a lot of people or I would not be the only one with a problem. b. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to integrate Asterisk with Avaya
Please ignore last message of mine. I was busy with doing multiple tasks here at work and I falsely thought this was a reply to a mail I sent just a while ago. :P -Original Message- From: Kyriakos [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 20, 2007 5:42 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] How to integrate Asterisk with Avaya Hi, Im not talking about connecting Asterisk with Avaya system. I just mentioned Avaya because on a presentation I've been to, they said that this could be done. I want to do this on Asterisk. I already have a call centre setup with 5 different queues with same weights but it seems that ACD is not taking into consideration how long the call has been waiting in each queue and its randomly choosing queue for next available agent. The result is that some calls might be answered even though they have much shorter waiting time in queue than calls to other queues. I would like to create only one queue and have customers choose using an IVR a call category (Sales, Tech support, accounting, etc.), tag the call with a code or something so asterisk will know what this call is about, send it into queue along with other category calls, and when its time to be sent to the next available agent, it should be sent to an agent marked with the same code that the call is tagged with ie if call is tagged for sales then the available agent with the highest weight for skill SALES should answer the calls. This way I make sure I have FIFO for incoming calls. BR KM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, November 20, 2007 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to integrate Asterisk with Avaya which Avaya system? and what are you trying to add with asterisk? On 11/20/07, Dovid B [EMAIL PROTECTED] wrote: Hello Everyone, Can someone please point to sources how to integrate Asterisk PBX with Avaya..? What normalize and expose protocol/API does Avaya support which can be use with Asterisk? Thanks in advance, -C What are you trying to do between asterisk and Avaya ? Avaya has software (forgot the name of it) that allows you to connect to it and set the configs etc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime extensions configuration - calling user filtering
I tried it with 1.4 and it didn't work with standard settings and no magic:) On Nov 20, 2007 4:32 PM, Ricardo Carvalho [EMAIL PROTECTED] wrote: As much I as can tell, Asterisk version 1.2 doesn't support the ex-girlfriend logic that you ask. I didn't test that feature with 1.4 releases, maybe they already implement it. Regards, Ricardo Carvalho.. On Nov 20, 2007 2:51 PM, Tomasz Zieleniewski [EMAIL PROTECTED] wrote: Hi, Is it possible to filter the calling user with the usage of mysql realtime the same as it is done in extensions.conf file: exten = some_exten/calling-user is there some flag which activates this extra check?? Cheers Tomasz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime extensions configuration - calling user filtering
If you really want to use some DB to read/write your dialplan, the best thing for you would be to write some scripts to generate text files from the contents of the tables of your DB. Those files can then be loaded in the extensions.conf file with the sentence: #include generated_file.txt. In the same script you can even do some asterisk -r -x extensions reload command, and then you'll have your own realtime extensions working with the ex-girlfriend logic you wanted! I implemented this way because I had the same problem as you... :) Regards, Ricardo Carvalho. On Nov 20, 2007 4:16 PM, Tomasz Zieleniewski [EMAIL PROTECTED] wrote: I tried it with 1.4 and it didn't work with standard settings and no magic:) On Nov 20, 2007 4:32 PM, Ricardo Carvalho [EMAIL PROTECTED] wrote: As much I as can tell, Asterisk version 1.2 doesn't support the ex-girlfriend logic that you ask. I didn't test that feature with 1.4 releases, maybe they already implement it. Regards, Ricardo Carvalho.. On Nov 20, 2007 2:51 PM, Tomasz Zieleniewski [EMAIL PROTECTED] wrote: Hi, Is it possible to filter the calling user with the usage of mysql realtime the same as it is done in extensions.conf file: exten = some_exten/calling-user is there some flag which activates this extra check?? Cheers Tomasz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reporting bugs
I recently subscribed to the bugs mailing list and submitted a suspected bug. The report seems to be ignored. I am guessing that it is being ignored because I am not actually an asterisk user and I am unable to supply the version or configuration of the suspect site. So then I thought I should get an account for one of the forums. I tried repeatedly to create an account but it always told me the image verification was incorrect. I think it was referring to the little picture with the letters displayed in it. I have created other accounts by this method and never encountered this difficulty. Two questions. How to get a report of a suspected bug to be taken seriously? How to get an account for one of the forums? I believe in open source software. I am trying to make a difference. Thanks, Rob ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: How to configure SIP domain on SPA942
Take a look at the admin guides at http://spc.pifiu.com On Nov 18, 2007 10:53 PM, Philip Prindeville [EMAIL PROTECTED] wrote: I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from astlinux to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming canreinvite=no realm=redfish-solutions.com domain=redfish-solutions.com,incoming-redfish tos=184 disallow=all allow=ulaw allow=gsm localnet=192.168.10.0/255.255.255.0 externip=X.X.X.X (Footnote: do I need a default context? I'd rather not having one... I'd rather specify where my calls go explicitly...) However, my phones don't seem to be registering with any (symbolic) domain... just the IP address of their DHCP or TFTP server (can't tell which, since it's the same box). -- SIP read from 192.168.10.187:5060: REGISTER sip:192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.187:5060;branch=z9hG4bK-1e31e66f From: sip:[EMAIL PROTECTED];tag=e798d04e1a8af3a6o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 58671 REGISTER Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;expires=3600 User-Agent: Linksys/SPA942-5.1.15(a) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces pbx2*CLI --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.10.187 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.10.187:5060: SIP/2.0 404 Not found (unknown domain) Via: SIP/2.0/UDP 192.168.10.187:5060;branch=z9hG4bK-1e31e66f;received=192.168.10.187 From: sip:[EMAIL PROTECTED];tag=e798d04e1a8af3a6o0 To: sip:[EMAIL PROTECTED];tag=as7c1c3fa2 Call-ID: [EMAIL PROTECTED] CSeq: 58671 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 The config seems to take: Our local SIP domains: Context Set by redfish-solutions.comincoming-redfish [Configured] So, what's the DHCP option (or the HTTP knob) to tweak to get the phones to think they are in the redfish-solutions.com domain? Thanks, -Philip ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reporting bugs
Hi On Tue, Nov 20, 2007 at 09:33:59AM -0800, Robert Dyck wrote: I recently subscribed to the bugs mailing list and submitted a suspected bug. I figure you refer to http://lists.digium.com/mailman/listinfo/asterisk-bugs . This list is not used by users to report bugs. It is used by the software of the bug tracker to report status to users. How to get a report of a suspected bug to be taken seriously? Report it in the proper place: http://bugs.digium.com/ See http://asterisk.org/developers/bug-guidelines . Sorry for the confusion. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reporting bugs
Robert Dyck wrote: Two questions. How to get a report of a suspected bug to be taken seriously? http://bugs.digium.com How to get an account for one of the forums? That I don't know, I've never used them. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXO Hangs up automatically
Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an Acer Machine On receiving an incoming call, Connected to Asterisk 1.2.17 currently running on pbx (pid = 5092) Verbosity was 16 and is now 22 -- Starting simple switch on 'Zap/4-1' Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:4502 __zt_exception: Exception on 16, channel 4 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:3687 zt_handle_event: Got event On hook(1) on channel 4 (index 0) Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1586 zt_disable_ec: disabled echo cancellation on channel 4 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:5683 ss_thread: waitfordigit returned 0... Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:2493 zt_hangup: Hangup: channel: 4 index = 0, normal = 16, callwait = -1, thirdcall = -1 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1586 zt_disable_ec: disabled echo cancellation on channel 4 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:2933 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/4-1 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1523 update_conf: Updated conferencing on 4, with 0 conference users -- Hungup 'Zap/4-1' pbx*CLI On Trying to make an outgoing call Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Match Found Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:10669 handle_request_invite: Checking SIP call limits for device 319 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:6267 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Nov 20 20:51:48 DEBUG[5101]: channel.c:775 channel_find_locked: Avoiding initial deadlock for 'SIP/319-081d8e00' -- Executing Dial(SIP/319-081d8e00, Zap/1/0004479086365389) in new stack Nov 20 20:51:48 DEBUG[6042]: chan_zap.c:2065 zt_call: Dialing '0004479086365389' Nov 20 20:51:48 DEBUG[6042]: chan_zap.c:2137 zt_call: Deferring dialing... -- Called 1/0752707099 Nov 20 20:51:49 DEBUG[6042]: chan_zap.c:4502 __zt_exception: Exception on 17, channel 1 Nov 20 20:51:49 DEBUG[6042]: chan_zap.c:3687 zt_handle_event: Got event Hook Transition Complete(12) on channel 1 (index 0) Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:4502 __zt_exception: Exception on 17, channel 1 Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:3687 zt_handle_event: Got event Dial Complete(9) on channel 1 (index 0) Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:1554 zt_enable_ec: Enabled echo cancellation on channel 1 -- Zap/1-1 answered SIP/319-081d8e00 Nov 20 20:51:51 DEBUG[5101]: channel.c:775 channel_find_locked: Avoiding initial deadlock for 'SIP/319-081d8e00' -- Limit Data for this call: -- - timelimit = 0 -- - play_warning = 0 -- - warning_sound = (null) Nov 20 20:51:51 DEBUG[5110]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 102: Match Found Nov 20 20:51:51 DEBUG[6042]: chan_sip.c:3051 sip_rtp_read: Oooh, format changed to 256 The Call doesn't go through --- Out put of `lspci` . . 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface . . . --- Output of `lsmod` Module Size Used by wctdm 37184 4 . . . - Output of /proc/zaptel/1 [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV H Board 1 1 WCTDM/0/0 FXSKS (In use) 2 WCTDM/0/1 FXOKS (In use) 3 WCTDM/0/2 FXOKS (In use) 4 WCTDM/0/3 FXOKS (In use) [EMAIL PROTECTED] ~]# Output of ztcfg - [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV H Board 1 1 WCTDM/0/0 FXSKS (In use) 2 WCTDM/0/1 FXOKS (In use) 3 WCTDM/0/2 FXOKS (In use) 4 WCTDM/0/3 FXOKS (In use) -- [EMAIL PROTECTED] ~]# ztcfg - Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) 4 channels configured. [EMAIL PROTECTED] ~]# My /etc/zaptel.conf [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf fxsks=1 fxoks=2-4 loadzone = us defaultzone=us [EMAIL PROTECTED] ~]# -- My /etc/asterisk/zapata.conf [EMAIL PROTECTED] ~]# cat /etc/asterisk/zapata.conf [channels] group=2 signalling=fxo_ks context=outgoing callerid=Extensions channel = 2-4 group=3 signalling=fxs_ks context=analog-incoming channel = 1 [EMAIL PROTECTED] ~]# Out put of zap show pbx*CLI zap
Re: [asterisk-users] [asterisk-dev] trunk working under windows!
Cool, i'll help out a bit with the windows port, i will start right away with a new project on asteriskguru making nightly executable builds and installers - will post the links in -users when i'm done. Well done luigi, this will make it a lot easier for a lot of non linux guys to make their first steps in the asterisk world Crossposted to -users. Zoa Luigi Rizzo wrote: As a result of the commit below, now trunk can be built and run under Windows/cygwin, including the building of modules. Haven't checked yet the functionality - some modules surely cause ill side effects or deadlocks on exit, so you need to play a bit with modules.conf . If you want to play with a very minimal version the following does something: ; -- modules.conf [modules] autoload=no load = res_monitor.so load = res_features.so load = chan_sip.so Unfortunately, loading other modules is a bit critical and depending on the order or the timing you get crashes etc. To build trunk under windows/cygwin you need at least the following pieces: bash binutils curl gcc libiconv minires (resolver library) libdb4.3(probably db4.2 too) and a bit of patience because the build takes around 15min or more. cheers luigi On Tue, Nov 20, 2007 at 04:12:11PM -, SVN commits to the Asterisk project wrote: Author: rizzo Date: Tue Nov 20 10:12:10 2007 New Revision: 89454 URL: http://svn.digium.com/view/asterisk?view=revrev=89454 Log: Fix building of modules under cygwin. After this commit we can actually load modules under windows, and we can start debugging more interesting problems related to the load order and functionality of modules. Modified: trunk/Makefile.moddir_rules trunk/apps/Makefile trunk/channels/Makefile trunk/pbx/Makefile trunk/res/Makefile Modified: trunk/Makefile.moddir_rules URL: http://svn.digium.com/view/asterisk/trunk/Makefile.moddir_rules?view=diffrev=89454r1=89453r2=89454 == --- trunk/Makefile.moddir_rules (original) +++ trunk/Makefile.moddir_rules Tue Nov 20 10:12:10 2007 @@ -66,9 +66,8 @@ ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) # linker options and extra libraries for cygwin SOLINK=-Wl,[EMAIL PROTECTED] -shared - LIBS+=-L../main -lasterisk -L../res + LIBS+=-L../main -lasterisk -L../res $([EMAIL PROTECTED]) # additional libraries in res/ - LIBS_RES:= -lres_monitor -lres_adsi -lres_features endif endif Modified: trunk/apps/Makefile URL: http://svn.digium.com/view/asterisk/trunk/apps/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/apps/Makefile (original) +++ trunk/apps/Makefile Tue Nov 20 10:12:10 2007 @@ -39,3 +39,9 @@ all: _all include $(ASTTOPDIR)/Makefile.moddir_rules + +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_features.so -lres_ael_share.so -lres_monitor.so -lres_speech.so + LIBS+= -lres_smdi.so +endif + Modified: trunk/channels/Makefile URL: http://svn.digium.com/view/asterisk/trunk/channels/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/channels/Makefile (original) +++ trunk/channels/Makefile Tue Nov 20 10:12:10 2007 @@ -64,6 +64,10 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_monitor.so -lres_features.so +endif + clean:: rm -f gentone $(MAKE) -C misdn clean Modified: trunk/pbx/Makefile URL: http://svn.digium.com/view/asterisk/trunk/pbx/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/pbx/Makefile (original) +++ trunk/pbx/Makefile Tue Nov 20 10:12:10 2007 @@ -25,6 +25,10 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_ael_share.so -lres_monitor.so +endif + clean:: rm -f ael/*.o Modified: trunk/res/Makefile URL: http://svn.digium.com/view/asterisk/trunk/res/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/res/Makefile (original) +++ trunk/res/Makefile Tue Nov 20 10:12:10 2007 @@ -25,6 +25,13 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + # order-only dependency: build res_monitor before res_features + res_features.so: | res_monitor.so + # res_features uses some functions from res_monitor + res_features.so_LIBS:= -lres_monitor.so +endif + ael/ael_lex.o: ael/ael_lex.c ../include/asterisk/ael_structs.h ael/ael.tab.h ael/ael_lex.o: ASTCFLAGS+=-I. -Iael
[asterisk-users] e911
One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: How to configure SIP domain on SPA942
Yeah, I looked at LinksysSPATFTPProv.pdf... It doesn't say, however, how to get the phone's configuration out as a flat XML file. Only how to push the file back into the phone. Nor does it say how the phone derives its SIP domain. -Philip [EMAIL PROTECTED] wrote: Take a look at the admin guides at http://spc.pifiu.com On Nov 18, 2007 10:53 PM, Philip Prindeville [EMAIL PROTECTED] wrote: I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from astlinux to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming canreinvite=no realm=redfish-solutions.com domain=redfish-solutions.com,incoming-redfish tos=184 disallow=all allow=ulaw allow=gsm localnet=192.168.10.0/255.255.255.0 externip=X.X.X.X (Footnote: do I need a default context? I'd rather not having one... I'd rather specify where my calls go explicitly...) However, my phones don't seem to be registering with any (symbolic) domain... just the IP address of their DHCP or TFTP server (can't tell which, since it's the same box). -- SIP read from 192.168.10.187:5060: REGISTER sip:192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.187:5060;branch=z9hG4bK-1e31e66f From: sip:[EMAIL PROTECTED];tag=e798d04e1a8af3a6o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 58671 REGISTER Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;expires=3600 User-Agent: Linksys/SPA942-5.1.15(a) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces pbx2*CLI --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.10.187 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.10.187:5060: SIP/2.0 404 Not found (unknown domain) Via: SIP/2.0/UDP 192.168.10.187:5060;branch=z9hG4bK-1e31e66f;received=192.168.10.187 From: sip:[EMAIL PROTECTED];tag=e798d04e1a8af3a6o0 To: sip:[EMAIL PROTECTED];tag=as7c1c3fa2 Call-ID: [EMAIL PROTECTED] CSeq: 58671 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 The config seems to take: Our local SIP domains: Context Set by redfish-solutions.comincoming-redfish [Configured] So, what's the DHCP option (or the HTTP knob) to tweak to get the phones to think they are in the redfish-solutions.com domain? Thanks, -Philip ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxpeers from Realtime
Hello asterisk users, here is a little problem pulling out iax peers from
real time database
I have the following peer configured in my database
mysql select
name,username,secret,type,context,host,disallow,allow,defaultip,deny,permit,
ipaddr,port from iax_users where name='iaxtermination';
++--+--+--+-
---+--+--+---+---+--+---
-++--+
| name | username | secret | type |
context| host | disallow | allow | defaultip | deny
| permit | ipaddr | port |
++--+--+--+-
---+--+--+---+---+--+---
-++--+
| iaxtermination | NULL | xx | peer |
iaxtermination | 195.66.85.55 | all | gsm;alaw;ulaw | NULL | NULL
| NULL | NULL |0 |
++--+--+--+-
---+--+--+---+---+--+---
-++--+
And the extension to use it is..
mysql select * from extensions_table where context='iaxtermination' order
by context,exten,priority;
+-++---+--+--+--
--+
| id | context| exten | priority | app | appdata
|
+-++---+--+--+--
--+
| 719 | iaxtermination | _*92. |1 | Dial |
IAX2/[EMAIL PROTECTED]/011${EXTEN:3} |
+-++---+--+--+--
--+
But whe I try to place the call I ge some wiered messages saying doing --
lookup for '195.66.85.55' ---
Here is de details output
-- Executing Dial(SIP/FJST1001-087b8138,
IAX2/[EMAIL PROTECTED]/01115141234123)
-- doing lookup for '195.66.85.55'
-- doing lookup for '195.66.85.55'
-- Called [EMAIL PROTECTED]/01115141234123
-- doing lookup for '195.66.85.55'
-- Hungup 'IAX2/iaxtermination-2'
== Spawn extension (autorized, *9215141234123, 1) exited non-zero on
'SIP/FJST1001-087b8138'
-- doing lookup for '195.66.85.55'
I know iax_users and sip_users on realtime are suposte to be dynamic, on an
incomming request, ARA will look on to the database and pull the information
necessary to authenticate the user and let them place calls, etc, and then
after hangup ARA will delete/remove the information about that particular
user, but this is for incomming requests, (users,friends) but what about
peers ?
if I issue a iax2 show peers command on the console, ther is absolutly no
informtion about the peer, see output...
lnxca*CLI iax2 show peers
Name/UsernameHost Mask Port Status
0 iax2 peers [0 online, 0 offline, 0 unmonitored]
A little test I did wast configuring the peer static and once the iax was
reloaded I was able to see peers information in the console and able to
successfull place a call
/etc/iax.conf
[iaxtermination]
type = peer
host = 195.66.85.55
secret = x
auth = md5
notransfer = yes
context = a2billing
disallow=all
allow=gsm
allow=alaw
allow=ulaw
lnxca*CLI iax2 show peers
Name/UsernameHost Mask Port Status
iaxtermination 195.66.85.55(S) 255.255.255.255 4569
Unmonitored
1 iax2 peers [0 online, 0 offline, 1 unmonitored]
-- Executing Dial(SIP/FJST1001-087b8138,
IAX2/[EMAIL PROTECTED]/01115146421231234)
-- Called [EMAIL PROTECTED]/01115146421231234
-- Call accepted by 195.66.85.55 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxtermination-7 is ringing
-- Hungup 'IAX2/iaxtermination-7'
Is there something missing on my peer's config on the iax_users table? here
is the structure for the iax_users table
mysql describe iax_users;
+-+--+--+-+-++
| Field | Type | Null | Key | Default | Extra |
+-+--+--+-+-++
| id | int(11) | NO | PRI | NULL| auto_increment |
| name| varchar(30) | NO | UNI | ||
| username| varchar(30) | YES | | NULL||
| type| varchar(6) | NO | | ||
| secret | varchar(50) | YES | | NULL||
| md5secret | varchar(32) | YES | | NULL||
| dbsecret| varchar(100) | YES | | NULL||
| notransfer | varchar(10) | YES | | NULL||
| inkeys | varchar(100) | YES | | NULL||
| auth| varchar(100) | YES | | NULL||
| accountcode | varchar(100) | YES | | NULL|
Re: [asterisk-users] FXO Hangs up automatically
On Tue, Nov 20, 2007 at 09:01:22PM +0300, Timothy Smith wrote: Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an Acer Machine On receiving an incoming call, Connected to Asterisk 1.2.17 currently running on pbx (pid = 5092) Verbosity was 16 and is now 22 -- Starting simple switch on 'Zap/4-1' Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:4502 __zt_exception: Exception on 16, channel 4 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:3687 zt_handle_event: Got event On hook(1) on channel 4 (index 0) Hmmm it is your side (not the remote side) that can initiate an On-Hook. Can you try it with a simple analog phone instead? Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1586 zt_disable_ec: disabled echo cancellation on channel 4 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:5683 ss_thread: waitfordigit returned 0... Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:2493 zt_hangup: Hangup: channel: 4 index = 0, normal = 16, callwait = -1, thirdcall = -1 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1586 zt_disable_ec: disabled echo cancellation on channel 4 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:2933 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/4-1 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1523 update_conf: Updated conferencing on 4, with 0 conference users -- Hungup 'Zap/4-1' pbx*CLI On Trying to make an outgoing call Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Match Found Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:10669 handle_request_invite: Checking SIP call limits for device 319 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:6267 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Nov 20 20:51:48 DEBUG[5101]: channel.c:775 channel_find_locked: Avoiding initial deadlock for 'SIP/319-081d8e00' -- Executing Dial(SIP/319-081d8e00, Zap/1/0004479086365389) in new stack Nov 20 20:51:48 DEBUG[6042]: chan_zap.c:2065 zt_call: Dialing '0004479086365389' Nov 20 20:51:48 DEBUG[6042]: chan_zap.c:2137 zt_call: Deferring dialing... -- Called 1/0752707099 Nov 20 20:51:49 DEBUG[6042]: chan_zap.c:4502 __zt_exception: Exception on 17, channel 1 Nov 20 20:51:49 DEBUG[6042]: chan_zap.c:3687 zt_handle_event: Got event Hook Transition Complete(12) on channel 1 (index 0) Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:4502 __zt_exception: Exception on 17, channel 1 Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:3687 zt_handle_event: Got event Dial Complete(9) on channel 1 (index 0) Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:1554 zt_enable_ec: Enabled echo cancellation on channel 1 -- Zap/1-1 answered SIP/319-081d8e00 Nov 20 20:51:51 DEBUG[5101]: channel.c:775 channel_find_locked: Avoiding initial deadlock for 'SIP/319-081d8e00' -- Limit Data for this call: -- - timelimit = 0 -- - play_warning = 0 -- - warning_sound = (null) Nov 20 20:51:51 DEBUG[5110]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 102: Match Found Nov 20 20:51:51 DEBUG[6042]: chan_sip.c:3051 sip_rtp_read: Oooh, format changed to 256 So what exactly is wrong here? SIP talking to Zap with a ulaw format. Anything wrong? The Call doesn't go through --- Out put of `lspci` . . 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface . . . --- Output of `lsmod` Module Size Used by wctdm 37184 4 . . . - Output of /proc/zaptel/1 [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV H Board 1 1 WCTDM/0/0 FXSKS (In use) 2 WCTDM/0/1 FXOKS (In use) 3 WCTDM/0/2 FXOKS (In use) 4 WCTDM/0/3 FXOKS (In use) [EMAIL PROTECTED] ~]# Output of ztcfg - [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV H Board 1 1 WCTDM/0/0 FXSKS (In use) 2 WCTDM/0/1 FXOKS (In use) 3 WCTDM/0/2 FXOKS (In use) 4 WCTDM/0/3 FXOKS (In use) -- [EMAIL PROTECTED] ~]# ztcfg - Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) 4 channels configured. [EMAIL PROTECTED] ~]# My /etc/zaptel.conf
[asterisk-users] automatic blind transfer calls
Hi, I would like to do a blind transfer in an automatic way. For example I dial 5 during a call and the caller is blind transferred to SIP/578 (for example). I saw that with features.conf it is not possible to do that. Regards gianrico ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Route an incoming call by ANI*DNIS
Thank you all,
It just so turns out that it was a bad zaptel module. We saw
another post on digiums site where someone was having the exact same
problem with several versions of zaptel. We changed to the one that he
said worked (1.2.21), and all is well now. (And asterisk is now parsing
the ani and dnis properly).
Tilghman Lesher wrote:
On Tuesday 20 November 2007 02:38:38 Atis Lezdins wrote:
I won't be able to help with hardware part, but there's a simple trick
to get them as you want:
[incoming]
_X.,1,Set(DNIS=${CUT(${EXTEN:-4})})
_X.,2,Goto,dnis,${DNIS},1
[dnis]
6789 = ...
I don't think you've actually tested this, because if you had, you would find
that it does not work.
[incoming]
exten = _X.,1,Goto(dnis,${EXTEN:-4},1)
[dnis]
exten = 6789,1,.
___
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[asterisk-users] Bugtracker to use with Asterisk?
Hello Now that I have my first IVR up and running, I'd like to have Asterisk create tickets in a bug tracker every time a call comes in. It's a nice way to know who's calling and why, before following up on the cause for the call. There are tons of bugtracking apps out there. Do you know of some that I should look at? Ideally, the interface shouldn't be much busier than JoS http://discuss.joelonsoftware.com/?joel . Thank you ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interface with NEC NEAX 2400
Is there anyone out there who has tried to connect up an asterisk box to make and take calls through a NEC NEAX 2400 using Q.sig or anything like it? Can anyone tell me if it is possible? Phil, I've successfully connected my NEAX 2400 to Asterisk using line side and trunk side T1's. I've only documented the line side setup: http://voip-info.org/wiki/index.php?page=Asterisk+NEAX2400+LineSide I've never tried using a PRI card though... HtH, MC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime extensions configuration - calling user filtering
On Tuesday 20 November 2007 11:18:45 Ricardo Carvalho wrote: If you really want to use some DB to read/write your dialplan, the best thing for you would be to write some scripts to generate text files from the contents of the tables of your DB. Those files can then be loaded in the extensions.conf file with the sentence: #include generated_file.txt. In the same script you can even do some asterisk -r -x extensions reload command, and then you'll have your own realtime extensions working with the ex-girlfriend logic you wanted! I implemented this way because I had the same problem as you... :) Or you could use func_odbc and get a real dynamic dialplan, instead of moving your static dialplan to a database, which really makes it no more dynamic. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco phones and 32 directory object limit
Hello List, For those of you with Cisco phones and XML directories and large user bases, how do you handle the 32 directory object limit? I know you can create multiple xml files with 32 objects in each but this just seems really sloppy. I would like to have one large directory. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] trunk working under windows!
but ... why? Zoa wrote: Cool, i'll help out a bit with the windows port, i will start right away with a new project on asteriskguru making nightly executable builds and installers - will post the links in -users when i'm done. Well done luigi, this will make it a lot easier for a lot of non linux guys to make their first steps in the asterisk world Crossposted to -users. Zoa Luigi Rizzo wrote: As a result of the commit below, now trunk can be built and run under Windows/cygwin, including the building of modules. Haven't checked yet the functionality - some modules surely cause ill side effects or deadlocks on exit, so you need to play a bit with modules.conf . If you want to play with a very minimal version the following does something: ; -- modules.conf [modules] autoload=no load = res_monitor.so load = res_features.so load = chan_sip.so Unfortunately, loading other modules is a bit critical and depending on the order or the timing you get crashes etc. To build trunk under windows/cygwin you need at least the following pieces: bash binutils curl gcc libiconv minires (resolver library) libdb4.3(probably db4.2 too) and a bit of patience because the build takes around 15min or more. cheers luigi On Tue, Nov 20, 2007 at 04:12:11PM -, SVN commits to the Asterisk project wrote: Author: rizzo Date: Tue Nov 20 10:12:10 2007 New Revision: 89454 URL: http://svn.digium.com/view/asterisk?view=revrev=89454 Log: Fix building of modules under cygwin. After this commit we can actually load modules under windows, and we can start debugging more interesting problems related to the load order and functionality of modules. Modified: trunk/Makefile.moddir_rules trunk/apps/Makefile trunk/channels/Makefile trunk/pbx/Makefile trunk/res/Makefile Modified: trunk/Makefile.moddir_rules URL: http://svn.digium.com/view/asterisk/trunk/Makefile.moddir_rules?view=diffrev=89454r1=89453r2=89454 == --- trunk/Makefile.moddir_rules (original) +++ trunk/Makefile.moddir_rules Tue Nov 20 10:12:10 2007 @@ -66,9 +66,8 @@ ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) # linker options and extra libraries for cygwin SOLINK=-Wl,[EMAIL PROTECTED] -shared - LIBS+=-L../main -lasterisk -L../res + LIBS+=-L../main -lasterisk -L../res $([EMAIL PROTECTED]) # additional libraries in res/ - LIBS_RES:= -lres_monitor -lres_adsi -lres_features endif endif Modified: trunk/apps/Makefile URL: http://svn.digium.com/view/asterisk/trunk/apps/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/apps/Makefile (original) +++ trunk/apps/Makefile Tue Nov 20 10:12:10 2007 @@ -39,3 +39,9 @@ all: _all include $(ASTTOPDIR)/Makefile.moddir_rules + +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_features.so -lres_ael_share.so -lres_monitor.so -lres_speech.so + LIBS+= -lres_smdi.so +endif + Modified: trunk/channels/Makefile URL: http://svn.digium.com/view/asterisk/trunk/channels/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/channels/Makefile (original) +++ trunk/channels/Makefile Tue Nov 20 10:12:10 2007 @@ -64,6 +64,10 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_monitor.so -lres_features.so +endif + clean:: rm -f gentone $(MAKE) -C misdn clean Modified: trunk/pbx/Makefile URL: http://svn.digium.com/view/asterisk/trunk/pbx/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/pbx/Makefile (original) +++ trunk/pbx/Makefile Tue Nov 20 10:12:10 2007 @@ -25,6 +25,10 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_ael_share.so -lres_monitor.so +endif + clean:: rm -f ael/*.o Modified: trunk/res/Makefile URL: http://svn.digium.com/view/asterisk/trunk/res/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/res/Makefile (original) +++ trunk/res/Makefile Tue Nov 20 10:12:10 2007 @@ -25,6 +25,13 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + # order-only dependency: build res_monitor before res_features + res_features.so: | res_monitor.so + # res_features uses some functions from res_monitor + res_features.so_LIBS:= -lres_monitor.so +endif + ael/ael_lex.o: ael/ael_lex.c ../include/asterisk/ael_structs.h ael/ael.tab.h ael/ael_lex.o:
[asterisk-users] FXO incomming call hangup problem
Dear all I have asterisk with TDM808B FXO port with i call in asterisk and i promt IVR then user dial extention for user then my SIP phone rining but i disconnect or hangup my mobile phone but still SIP phone rining and stop rining after timeout is there any PSTN problme or FXO signalling problme i have configuraed singalling=fxs_ks PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org - Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to receive manager events from commands made by an AGI script?
Hi all, I'm new on this list, my name is Noel. :D I developed a system using AGI and now I'm trying to develop a system that listen events fired by Manager API. I have realized that I don't receive events from commands made by an AGI script like play a file or record a file. Is there a way to receive such events? Sorry about the poor English. Thanks in advance. Noel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Pcengines Alix board
Il giorno Mon, 19 Nov 2007 08:54:38 -0500 Matthew Rubenstein [EMAIL PROTECTED] ha scritto: Other than the Alix board, what else is needed to make a working PC? You need a CF as main storage device (it is mounted ro on /). I also use an USB stick where I mount /var in rw mode. Obviously you need even a power supply (sold by Pcengines). Giuseppe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Pcengines Alix board
Date: Mon, 19 Nov 2007 10:39:31 -0600 From: Bob Pierce [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk on Pcengines Alix board To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain On Sun, 2007-11-18 at 22:14 +0100, Giuseppe Barichello wrote: I have successfully compiled and installed Asterisk on an Alix board (AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian variant). I'm using it at home for a month. That's very interesting! I've been curious about trying this. Did you run across any challenges getting this setup? Two main issues: 1) Understanding how voyage linux configures read-only and rw mounts (I wanted to mount all /var tree as rw) 2) Getting MOH play MP3 sound files with Debian standard packages: I had to recompile Asterisk from source to fix it. Giuseppe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to receive manager events from commands made by an AGI script?
That's because no event is being generated. I can do a quick patch for you and post it in mantis in order to accomplish that. But I am interested in know why you want to receive those events. I am in the middle of creating a new AGI application. As you probably know, you can launch AGI like this: AGI(agi://host ...) to execute AGI through a TCP socket instead of forking a new prcess in the local machine as AGI(script.php) does. What I am doing is a new way of executing AGI, where you will specify AGI(agi:async), which means, AGI commands will arrive asynchronously via the manager interface or the command line. Something like: Action: AGI Command: EXEC Playback Hello World Channel: SIP/23 (this channel must be in AGI(agi:async)) Or CLI AGI execute SIP/23 EXEC Playback Hello World This is sort of a plus, my initial intention is being able to execute AGI from the manager interface to control everything from the manager. Ahhh I just hijacked your post to write mi thoughts :( sorry about that. Let me know if you are interested in the patch to send events for each agi command executed. Warm Regards, - Moy On Nov 20, 2007 2:23 PM, Noel R. Morais [EMAIL PROTECTED] wrote: Hi all, I'm new on this list, my name is Noel. :D I developed a system using AGI and now I'm trying to develop a system that listen events fired by Manager API. I have realized that I don't receive events from commands made by an AGI script like play a file or record a file. Is there a way to receive such events? Sorry about the poor English. Thanks in advance. Noel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on Hold Problem w/ Transfers
I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. I've looked at the sample config files for 1.4 and nothing seems to jump out at me as to what the problem could be. For the purposes of figuring this out, I'm using Zaptel 1.4.6 for both 1.2and 1.4. Any clues? Thanks! -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration problem: UA - SER - Asterisk
Stefano, It is not Asterisk, It is SER (dispatcher module ?). Why Asterisk is acting as Register ? make sense use openSER as Register/Proxy and Asterisk only Proxy and MG Regards, Giovanni 2007/11/19, Stefano Capitanio [EMAIL PROTECTED]: Hi, we a have a SER (OpenSER) in front of 2 real-time Asterisk. SER simply forward SIP messages to 1 of the Asterisks: UA -- SER -- Asterisk We have a problem with REGISTERs: Asterisk answers with 200 OK, but changes the Contact header, inserting the IP of SER instead of the original IP (the IP of the UA). It seems that performs a sort of NAT-traversal, but all the elements are on public IPs! The Asterisk's version is 1.2.21, they are in real-time configuration, installed on a virtual machine with gentoo-linux. I've tried the same scenario with an Asterisk 1.0.9 (without virtual and without real-time) on a Fedora Core distribution, and it works. Any idea? Best regards, -Stefano ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold Problem w/ Transfers
On Tue, 2007-11-20 at 15:52 -0600, Lacy Moore wrote: I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. FWIW, I'm using 1.4.10 and music on hold for transfers is working fine... for as long as the tranferee remains on hold before the transfer process hangs up their channel that is. I wish I could figure that one out. :-) Not even sure how to debug it in fact. :-( b. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-Users: Termination
I wanted to see if anyone has set up a large amount of out bound only voip channels? We run analog autodialers connected to analog to voip gateways (dialogic boards to audiocodes mp-124's) Bandwidth and routing is being provided by a bonded t-1 (3 meg) cisco 2600 with 2 t1 cards and a 16 port netgear switch. My question ( if you can picture the setup) is if anyone can see a problem with the set-up I have described. There is no firewall or access list on the router. Just wide open internet. I have been running about 80 channels for over a year and my numbers have been down and I cannot tell if there are any problems Mark Adams Infinity Marketing Inc. 1-800-430-1478 Main 530-579-8856 Fax 216-441-4319 Tech Support ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold Problem w/ Transfers
On Nov 20, 2007 3:52 PM, Lacy Moore [EMAIL PROTECTED] wrote: I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. After posting, I dialed my cellphone, and music on hold works in all situations. It's something having to do with internal calls. I don't really care if that isn't working. I didn't think to try that first. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-Users: Termination
We are using only voip chanels with 400-500 channels. Although we are still in begining phase but i have not seen any problem as such. Thanks, Vivek On 11/20/07, Mark Adams [EMAIL PROTECTED] wrote: I wanted to see if anyone has set up a large amount of out bound only voip channels? We run analog autodialers connected to analog to voip gateways (dialogic boards to audiocodes mp-124's) Bandwidth and routing is being provided by a bonded t-1 (3 meg) cisco 2600 with 2 t1 cards and a 16 port netgear switch. My question ( if you can picture the setup) is if anyone can see a problem with the set-up I have described. There is no firewall or access list on the router. Just wide open internet. I have been running about 80 channels for over a year and my numbers have been down and I cannot tell if there are any problems Mark Adams Infinity Marketing Inc. 1-800-430-1478 Main 530-579-8856 Fax 216-441-4319 Tech Support ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Building an Asterisk 1.4 RPM
I'm a little confused. I'd like to build an RPM for Asterisk 1.4. Is it better to modify and use the spec file under redhat/asterisk.spec and run a 'make rpm', OR is it better to build a custom spec file from scratch and use 'rpmbuid -ba' specfile? How do people normally do it? The problem I see with a custom spec file is that since the source is all contained within a tar.gz file, there's no way to interactively run a 'make menuselect' first and customise or remove what you don't need. For example, if I don't do this, the ogg vorbis module is installed by default, and then when I go to install my rpm, there's complaints all round if the ogg vorbis libs aren't already installed. Doug. Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building an Asterisk 1.4 RPM
http://www.voip-info.org/tiki-index.php?page=Asterisk%20RPM Try this link. There is a lot of info and source rpms that you can rebuild. Jonn _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Tuesday, November 20, 2007 6:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Building an Asterisk 1.4 RPM I'm a little confused. I'd like to build an RPM for Asterisk 1.4. Is it better to modify and use the spec file under redhat/asterisk.spec and run a 'make rpm', OR is it better to build a custom spec file from scratch and use 'rpmbuid -ba' specfile? How do people normally do it? The problem I see with a custom spec file is that since the source is all contained within a tar.gz file, there's no way to interactively run a 'make menuselect' first and customise or remove what you don't need. For example, if I don't do this, the ogg vorbis module is installed by default, and then when I go to install my rpm, there's complaints all round if the ogg vorbis libs aren't already installed. Doug. _ Never miss a thing. Make Yahoo http://us.rd.yahoo.com/evt=51438/*http:/www.yahoo.com/r/hs your homepage. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quality after call transfer
Hi, We are using attended call transfer to transfer the call. In the direct call, the quality of the voice and dtmf are acceptable. After transfer, the quality becomes worst. Voice can't be heard clearly and dtmf wrong detection will occur sometime. I wonder call transfer will affect he quality of the call. Anyone has same experience? Anything to do in asterisk level can get a better quality after call transfer? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [1.4 - Record] How to tell if user did leave a msg?
Hello
I didn't find the answer in the ATOF 2nd Ed: When using the Record()
application, I need to know how it ended: Did the user leave a
message, or did he hang up?
If the latter, Asterisk stops right there, while I need to run some
other commands before hanging up:
exten = _[1-4],n,Playback(/root/asterisk_sound_files/leave_msg)
exten = _[1-4],n,Set(CALLTIME=${STRFTIME(${EPOCH},,%d-%b-%Y-%Hh%M)})
;check if left message : if nothing, script ends there!
exten = _[1-4],n,Record(/tmp/${CALLTIME}.wav,3,30)
exten = _[1-4],n,TrySystem(mv /tmp/${CALLTIME}.wav
/srv/www/lighttpd/asterisk)
exten = _[1-4],n,ExecIf($[${SYSTEMSTATUS}!= SUCCESS],Verbose,Failed
moving WAV file)
exten = _[1-4],n,TrySystem(/root/asterisk/send_call_notification.py
${CALLERIDNAME} ${CALLERIDNUM} ${SOFTWARE} ${CALLTIME}.wav)
exten = _[1-4],n,ExecIf($[${SYSTEMSTATUS} != SUCCESS],Verbose,Error
sending e-mail)
exten = _[1-4],n,Playback(/root/asterisk_sound_files/bye_bye)
exten = _[1-4],n,Hangup()
Should I use another application?
Thank you.
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Re: [asterisk-users] [1.4 - Record] How to tell if user did leave a msg?
page 511
use dialplan function STAT()
--
On Nov 20, 2007 9:42 PM, Vincent wrote:
Hello
I didn't find the answer in the ATOF 2nd Ed: When using the Record()
application, I need to know how it ended: Did the user leave a
message, or did he hang up?
If the latter, Asterisk stops right there, while I need to run some
other commands before hanging up:
exten = _[1-4],n,Playback(/root/asterisk_sound_files/leave_msg)
exten = _[1-4],n,Set(CALLTIME=${STRFTIME(${EPOCH},,%d-%b-%Y-%Hh%M)})
;check if left message : if nothing, script ends there!
exten = _[1-4],n,Record(/tmp/${CALLTIME}.wav,3,30)
exten = _[1-4],n,TrySystem(mv /tmp/${CALLTIME}.wav
/srv/www/lighttpd/asterisk)
exten = _[1-4],n,ExecIf($[${SYSTEMSTATUS}!= SUCCESS],Verbose,Failed
moving WAV file)
exten = _[1-4],n,TrySystem(/root/asterisk/send_call_notification.py
${CALLERIDNAME} ${CALLERIDNUM} ${SOFTWARE} ${CALLTIME}.wav)
exten = _[1-4],n,ExecIf($[${SYSTEMSTATUS} != SUCCESS],Verbose,Error
sending e-mail)
exten = _[1-4],n,Playback(/root/asterisk_sound_files/bye_bye)
exten = _[1-4],n,Hangup()
Should I use another application?
Thank you.
___
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[asterisk-users] Zaptel 1.4 spec file
Does anyone know where I can get an rpm spec file for zaptel 1.4.x? Thanks, Doug. Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. http://mobile.yahoo.com/sports;_ylt=At9_qDKvtAbMuh1G1SQtBI7ntAcJ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
