Re: [asterisk-users] Codec Negotiation problem
Hi Matt Thanks for your response. I have tried with two GXV3175 with same result. Let me dig deep on this to find out the route cause Sam Matthew Jordan wrote: > On Thu, Jun 13, 2013 at 12:04 PM, wrote: > >> Hi there >> >> I have asterisk 10.11.1 which seems to have problem negotiating codec. >> >> Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p >> and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw, >> h263p. I have tried similar combination of codecs and SIP phone but when >> making a video call, it report "Peer doesn't provide video". It seems >> Asterisk is failing to set capability correct. Both codecs are enabled >> on >> the SIP Phones >> >> > > > The 200 OK response from the called XLite phone is declining the video > stream: > > <--- SIP read from UDP:10.10.10.129:48464 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK368135b0;rport=5060 > Contact: > To: "SAM";tag=0c90cc0c > From: ;tag=as24914503 > Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA > CSeq: 102 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO > Content-Type: application/sdp > Supported: replaces, eventlist > User-Agent: X-Lite release 4.5.2 stamp 70142 > Content-Length: 234 > > v=0 > o=- 13015615910543193 2 IN IP4 10.10.10.129 > s=X-Lite 4 release 4.5.2 stamp 70142 > c=IN IP4 10.10.10.129 > t=0 0 > m=audio 53188 RTP/AVP 8 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > m=video 0 RTP/AVP 115 > <-> > --- (12 headers 10 lines) --- > Found RTP audio format 8 > Found RTP audio format 101 > Found audio description format telephone-event for ID 101 > Capabilities: us - (alaw|h263p), peer - > audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) > > Note that the port for the video stream is set to 0. > > Asterisk is doing the correct thing: it notes that the answer to its offer > declined the video stream, so it disables video for the call between the > two endpoints. > > Matt > > -- > Matthew Jordan > Digium, Inc. | Engineering Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec Negotiation problem
Hi there I have asterisk 10.11.1 which seems to have problem negotiating codec. Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw, h263p. I have tried similar combination of codecs and SIP phone but when making a video call, it report "Peer doesn't provide video". It seems Asterisk is failing to set capability correct. Both codecs are enabled on the SIP Phones --- (12 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|h263p), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.10.10.129:53188 Peer doesn't provide video Here is a sip show peer output and log when making calls. localhost*CLI> sip show peer 1003 * Name : 1003 Description : Secret : MD5Secret: Remote Secret: Context : video-users Subscr.Cont. : Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : 1003@device VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "device" <1003> MaxCallBR: 384 kbps Expire : 3605 Insecure : no Force rport : Yes ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 10.10.10.129:48464 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 1003 SIP Options : (none) Codecs : (alaw|h263p) Codec Order : (alaw:20,h263p:0) Auto-Framing : No Status : OK (8 ms) Useragent: X-Lite release 4.5.2 stamp 70142 Reg. Contact : sip:1003@10.10.10.129:48464;rinstance=cf0c3558f05c89dc Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No localhost*CLI> sip show peer 1004 * Name : 1004 Description : Secret : MD5Secret: Remote Secret: Context : video-users Subscr.Cont. : Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : 1004@device VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "device" <1004> MaxCallBR: 384 kbps Expire : 893 Insecure : no Force rport : Yes ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 10.10.10.107:21769 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 1004 SIP Options : (none) Codecs : (alaw|h263p) Codec Order : (alaw:20,h263p:0) Auto-Framing : No Status : OK (2 ms) Useragent: Grandstream GXV3175v2 1.0.1.19 Reg. Contact : sip:1004@10.10.10.107:21769 Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No localhost*CLI> <-> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.10.10.129:48464 ---> INVITE sip:1004@10.10.10.105 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.129:48464;branch=z9hG4bK-d8754z-25f65c322686d22e-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: "SAM";tag=0c90cc0c Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite release 4.5.2 stamp 70142 Authorization: Digest username="1003",realm="10.10.10.105",nonce="05e8af6e",uri="sip:1004@10.10.10.105",response="20e63a04aa86d6ec1d1e045c05159b39",algorithm=MD5 Content-Length: 418 v=0 o=- 13015615910543193 1 IN IP4 10.10.10.129 s=X-Lite 4 release 4.5.2 stamp 70142 c=IN IP4 10.10.10.129 t=0 0 m=audio 53188 RTP/AVP 8 0 101 a=rtpmap:101 tel
Re: [asterisk-users] Who said asterisk is not to the task
Hi Markus Quad core running of 4 physical processor machine, HP DL580G5 Sam Markus wrote: > Am 29.09.2012 10:49, schrieb resea...@businesstz.com: >> [tz-ivr01 ~]# uptime >> 11:00:32 up 776 days, 10:49, 3 users, load average: 3.06, 3.05, 2.57 >> Sharing is caring > > Is that a Quad Core CPU in your box? > > PS: Yes, Asterisk is great. :) > > > > > > > > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Who said asterisk is not to the task
[tz-ivr01 ~]# uptime 11:00:32 up 776 days, 10:49, 3 users, load average: 3.06, 3.05, 2.57 Sharing is caring [tz-ivr01 ~]# asterisk -rx 'core show channels' |wc -l 213 mysql> select count(*) from cdr where calldate > '2012-01-01 00:00:00' and calldate <'2012-09-29 00:00:00' group by disposition; +--+ | count(*) | +--+ | 42926974 | +--+ 1 row in set (1.63 sec) mysql> select disposition, sum(billsec) from cdr where calldate >'2012-01-01 00:00:00' and calldate <'2012-09-29 00:00:00' group by disposition; +-+--+ | disposition | sum(billsec) | +-+--+ | ANSWERED| 4262026740 | +-+--+ 1 row in set (1.21 sec) Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2GB Elastix memory limit
64bit has resolved my issue thax Alex Villacís Lasso wrote: > El 28/06/12 03:58, resea...@businesstz.com escribió: >> I have sevaral elastix installed but all of them show the physical >> memory >> is 2GB while the server has 4GB and some has 8GB. I've upgraded to PAE >> kernel but yet i cant see mem beyond 2GB. How can i configure the centos >> kernel to use more memory as the server is multipurpose >> > I think you should use the Elastix mailing lists for this question. But > you should try using the 64-bit Elastix instead. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2GB Elastix memory limit
I have sevaral elastix installed but all of them show the physical memory is 2GB while the server has 4GB and some has 8GB. I've upgraded to PAE kernel but yet i cant see mem beyond 2GB. How can i configure the centos kernel to use more memory as the server is multipurpose Thanks Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting Mac Address on connected IP phones
James Sharp wrote: > On 3/13/12 5:53 PM, Danny Nicholas wrote: >> Ping the phones, then run arp. >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of >> resea...@businesstz.com >> Sent: Tuesday, March 13, 2012 4:52 PM >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] Getting Mac Address on connected IP phones >> >> I am struggling to get the mac-addresses of IP phones that are connected >> to >> asterisk as the phone are in different VLAN with * and they were >> manually >> configured. I want to centralize their configuration using res_phoneprov >> or >> tftp >> >> I have tried nmap and arp in vain. >> >> Any idea? >> > > ping + arp isn't going to work if they're on a different VLAN. > I believe this will work: > > 1) Set up your TFTP server, but do not put any configuration files in > the /tftpboot directory (or whatever the directory is). > 2) Set the DHCP server on the phones' network to hand out the TFTP > server address. > 3) Reboot the phones > 4) Watch the TFTP server logs and you should see each phone request a > file based on its MAC. With no downloaded config file, the phone should > revert to what it already has in nvram. > 5) Collect MAC addresses out of the server logs > 6) Profit? > Handy but working plan. Let me give it a try Thanks Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting Mac Address on connected IP phones
I am struggling to get the mac-addresses of IP phones that are connected to asterisk as the phone are in different VLAN with * and they were manually configured. I want to centralize their configuration using res_phoneprov or tftp I have tried nmap and arp in vain. Any idea? Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force sip peers to re register
As Kevin pointed out, it is obvious that there is no way of remote reset those phones since their registration status are unknown. SIP NOTIFY will only attempt to consult a registered phone and therefore no need, should it be that way Let me reconsult polyocm guide and see if there is a quicker way as Eric mentioned Sam > On Mon, Mar 5, 2012 at 8:55 AM, Kevin P. Fleming > wrote: >> As Alex pointed out, if the Asterisk server in question needs the phones >> to >> re-register in order to send them calls, then it probably cannot send >> them >> SIP NOTIFY requests either. > > This. I don't see how it would be possible to tell the phones to > reboot unless you sent it from the server they are *currently* > registered to. And if you can do that...you don't need to do that... > >> In addition, this NOTIFY request does not cause a Polycom phone to >> "reset". >> It instructs the phone to check its provisioning server for any changes >> to >> its configuration, and if there are any then apply them (rebooting if >> necessary). If the configuration has not changed, sending the phone a >> check-conf NOTIFY should be a no-op. > > Make a small script that uses the touch command to update the > Polycom's config file mod time/date. Then issue the standard CLI > command for them to check config. No need to actually modify the > file, it just looks at date/time. > > -- > Carlos Alvarez > TelEvolve > 602-889-3003 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Force sip peers to re register
I have hundreds of sip endpoints (mostly polycom) which i would like to immediate request them to reregister when we failover/fallback to the standby server. However it takes so long and i would like to know if there is a command to force all sip peers to attempt registration. I have tried both 'service asterisk restart' and 'reload' in vain. IP phones can be accessed at that time but no registration happen. Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Sangoma A104 and euroisdn pri
Can you post outputs for the following commands;
#asterisk -rx 'pri show spans'
#asterisk -rx 'zap show channels'
#wanpipemon -i w1g1 -c Ta
Sam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius
Sent: Thursday, April 01, 2010 4:15 PM
To: Asterisk
Subject: [asterisk-users] Problem with Sangoma A104 and euroisdn pri
Hi all,
My problem boils down to these errors:
... Unable to create channel of type 'ZAP' (cause 34 -
Circuit/channel congestion)
== Everyone is busy/congested at this time
This is triggered by lines in extentions.conf such as:
exten => _X.,1,Dial(ZAP/g1/${EXTEN},,W)
The system is CentOS v5.2 with Asterisk 1.4.23
(druid-asterisk-1.4.23.1-2), a Sangoma A104 4-port card, Wanpipe
v3.4.4 and Zaptel v1.4.12.1. The system is attached to a single
EuroISDN PRI and is located in the Netherlands.
Besides the above error, I also noticed this:
CLI> pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T309 Timer: -1
T313 Timer: 4000
N200 Counter: 3
The status needs to be "Provisioned, Up, Active."
Following Sangoma's instructions for debugging an Asterisk PRI span, I
can confirm that there are only outgoing frames and that the D-channel
messages in Asterisk are the same as what the Wanpipe drivers are
seeing. So, assuming that my local telco (KPN Telecom) has activated
the D-channel, what else could possibly be causing this problem?
Thanks,
Jaap
PS -- Below are my current configuration files and debugging output:
==begin zaptel.conf
loadzone=us
defaultzone=us
span=1,0,0,ccs,hdb3
bchan=1-15,17-31
hardhdlc=16
==end zaptel.conf ==
==begin wanpipe1.conf ==
[devices]
wanpipe1 = WAN_AFT_TE1, Comment
[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment
[wanpipe1]
CARD_TYPE = AFT
S514CPU = A
CommPort = PRI
AUTO_PCISLOT = NO
PCISLOT = 4
PCIBUS = 13
FE_MEDIA= E1
FE_LCODE= HDB3
FE_FRAME= NCRC4
FE_LINE= 1
TE_CLOCK = NORMAL
TE_REF_CLOCK= 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE= NO
LBO = 120OH
FE_TXTRISTATE= NO
MTU = 1500
UDPPORT = 9000
TTL= 255
IGNORE_FRONT_END = NO
TDMV_SPAN= 1
TDMV_DCHAN= 16
TDMV_HW_DTMF= NO
TDMV_HW_FAX_DETECT = NO
[w1g1]
ACTIVE_CH= ALL
TDMV_HWEC= NO
==end wanpipe1.conf
==begin zapata.conf
[trunkgroups]
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
switchtype=euroisdn
context=default
group=1
signalling=pri_cpe
channel =>1-15,17-31
==end zapata.conf ==
Here's some debugging output:
=== begin debug info ==
# ztcfg -vv
Zaptel Version: 1.4.12.1
Echo Canceller: MG2
Configuration
==
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: Hardware assisted D-channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: Clear channel (Default) (Slaves: 24)
Channel 25: Clear channel (Default) (Slaves: 25)
Channel 26: Clear channel (Default) (Slaves: 26)
Channel 27: Clear channel (Default) (Slaves: 27)
Channel 28: Clear channel (Default) (Slaves: 28)
Chann
[asterisk-users] High Availability Asterisk PBX
Hi I have the following scenario A. A PBX on location A with network 192.168.1.1 with extension range 1XXX and connected to the PSTN Network via the E1 B. Another PBX on location B with network 172.30.18.1 with extension range 2XXX and connected to the PSTN Network via the E1 I need to configure the system and the endpoints such that when one system, says, A goes down, the system B assumes A responsibility. HALinux would have been my answer but this should work only on the same subnet Any advice Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER
Hi there I remember to ask this question in the past but now I have thought of something little bit difference. While I understand that asterisk dialplan accept the call to be answered[ Answer() ] in the dialplan, I wanna know if this is possible; i. A call on legacy PBX, extension to extension is made. ii. On call bridging, the legacy PBX initiate a third bridging to the recording system via an ISDN interface. iii. Conversation on Legacy continue but asterisk record this call until hangup is issued Please advice if this is possible. Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
>
> snip
>>>
>>
>> You are correct. 8 span which process up to 240 calls at pick time
>>
>>> If the system is actually performing fine then I'd just say that there
>> is something about the Asterisk threads that makes them look runnable
>> and that
>>> accounts for the high load average. ?Is the IVR an agi or fastagi or
>> what? -
>>
>> I have the agi scripts not as ivr but to help populate the required
>> information into mysql db. Probably here is where the problem lies i
>> have
>> to connect and disconnect to mysql each time a call is made or a
>> specific
>> menu is selected
>>
>> Here is the script
>> *
>> #!/usr/bin/perl -w
>> use strict;
>> use DBI();
>> use Scalar::Util qw/weaken/;
>>
>> my $cdr_log_file = "/var/log/asterisk/ivr_log";
>> my $mysql_host = "cdr01";
>> my $mysql_db = "ivrcdrdb";
>> my $mysql_table = "tbl_ivrcdr_details";
>> my $mysql_user = "ivruser";
>> my $mysql_pwd = "a09876a";
>>
>>
>> my $sth;
>>
>> my $data0= $ARGV[0];
>> my $data1= $ARGV[1];
>> my $data2= $ARGV[2];
>> my $data3= $ARGV[3];
>> my $data4= $ARGV[4];
>> my $data5= $ARGV[5];
>> my $data6= $ARGV[6];
>> my $data7= $ARGV[7];
>>
>>
>> # Connect to database
>> # print "Connecting to database...\n\n";
>> my $dbh =
>>
DBI->connect("DBI:mysql:database=$mysql_db;host=$mysql_host","$mysql_user","
$mysql_pwd",{'RaiseError'
>> => 1});
>>
>> my $insert_str = "insert into $mysql_table (calldate, language, src,
>> duration, accountcode, uniqueid, currentmenu, nextmenu) values
>> (\"$data0\", \"$data1\", \"$data2\", \"$data3\", ?\"$data4\",
>> \"$data5\",
>> \"$data6\", \"$data7\");\n";
>> ? ? ? $sth = $dbh->prepare($insert_str);
>> ? ? ? $sth->execute();
>>
>> # print "\n\nOK.\n";
>>
>> $sth->finish();
>> $dbh->disconnect();
>>
>>
>> # Trying to resolve memory leak should it happen
>> delete($ARGV[0]);
>> delete($ARGV[1]);
>> delete($ARGV[2]);
>> delete($ARGV[3]);
>> delete($ARGV[4]);
>> delete($ARGV[5]);
>> delete($ARGV[6]);
>> delete($ARGV[7]);
>>
>>
>> exit;
>> *
>>
>>> the code path may have a "spinlock" logic to it that means that many
>> threads
>>> are runnable but when scheduled just go back to sleep. ?That would
>> account for high load average with lots of spare CPU. ?If that's what is
>> happening then I wouldn't worry much more about it.
>>>
>>> Regards,
>>> Steve
>>
>> Regards
>> Sam
>
> If I were you, and I am not and never will be, I would move over to
> fastagi and offload all that Perl and database stuff off to a
> designated server just to handle that stuff.
>
> I have had the EXACT same problem and that is how it was fixed,
> fastagi running to a Windows box that had a process developed (written
> in C something) by the M$ developers to hit the M$SQL databases.
>
> We were also doing a ton of things with the AMI which we figured out
> how to do the same end result without banging on the AMI, such as
> using call files rather than AMI to originate a call.
>
> Load avg dropped to one or under if I remember correctly.
>
> Thanks,
> Steve Totaro
>
Thank you Steve for your recommendation. Ofcoz i have separate server that
is hosting the db and i will consider doing fastagi and see it it will help
@Phil. The credintials displayed there are dummy, so don't worry unless you
mean something else
@Steve Edward. Can you share your C agi codes? I presume what you want me to
do is rewrite the script in C and use it as compiled binary
@Tzafrir. How about this
[ivr4 ~]# ps aux | grep D
USER PID %CPU %MEMVSZ RSS TTY STAT START TIME COMMAND
root 1975 0.0 0.0 3920 688 pts/4S+ 13:17 0:00 grep D
root 3413 0.0 0.0 1832 576 ?Ss2009 80:58
/usr/sbin/mDNSResponder -b -f /etc/services_mDNS
I have killed that process but no changes
@All, looks like the conclusion has been made that this is to do with AGI.
Let me address it and see how it reacts. I shall feedback
Thanks
Sam
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[asterisk-users] Asterisk as the recording server for Avaya Definity
Has anyone tried to replace Witness or Nice recorder with asterisk. I saw a nice article on voip-info.org on how to replace voicemail server for Avaya Definity with asterisk. The idea behind is to record not only the external channels but also extension to extension (three way calling for which the third leg is asterisk PRI will do) Any suggestion will highly help Sam ___ -- 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] Unstable PRI interface: Link restart after few min::
Hello Team I have connected * running centos 5.2, asterisk 1.6.1 dahdi 2.1 to the telco but the link is very unstable (D-Channel restart after some few min) Below please find part of 'pri intensive debug span 2' for your advice. Looks like telco is sending disconnect request but cant establish reason for this > Supervisory frame: > SAPI: 00 C/R: 0 EA: 0 > TEI: 000EA: 1 > Zero: 0 S: 0 01: 1 [ RR (receive ready) ] > N(R): 002 P/F: 1 > 0 bytes of data INV-VOICESW01*CLI> < [ 00 01 01 05 ] < Supervisory frame: < SAPI: 00 C/R: 0 EA: 0 < TEI: 000EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 002 P/F: 1 < 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 1 to (but not including) 2 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 timer NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active q931.c:3015 q931_disconnect: call 6321 on channel 1 enters state 11 (Disconnect Request) > [ 00 01 04 04 08 02 98 b1 45 08 02 81 90 ] > Informational frame: > SAPI: 00 C/R: 0 EA: 0 > TEI: 000EA: 1 > N(S): 002 0: 0 > N(R): 002 P: 0 > 9 bytes of data Stopping T_203 timer Starting T_200 timer -- Restarting T200 timer > Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 6321/0x18B1) (Terminator) > Message type: DISCONNECT (69) > [08 02 81 90] > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) > Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Destroying the call, ourstate Disconnect Request, peerstate Disconnect Indication -- Hungup 'DAHDI/1-1' == Spawn extension (from-outside, 0222112211, 3) exited non-zero on 'DAHDI/32-1' == End MixMonitor Recording DAHDI/32-1 -- Hungup 'DAHDI/32-1' -- T200 counter expired, What to do... -- Retransmitting 13 bytes > [ 00 01 04 05 08 02 98 b1 45 08 02 81 90 ] > Informational frame: > SAPI: 00 C/R: 0 EA: 0 > TEI: 000EA: 1 > N(S): 002 0: 0 > N(R): 002 P: 1 > 9 bytes of data -- Rescheduling retransmission (1) INV-VOICESW01*CLI> < [ 00 01 01 07 ] < Supervisory frame: < SAPI: 00 C/R: 0 EA: 0 < TEI: 000EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 003 P/F: 1 < 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 1 to (but not including) 3 -- ACKing packet 2, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 timer INV-VOICESW01*CLI> < [ 02 01 04 06 08 02 18 b1 4d ] < Informational frame: < SAPI: 00 C/R: 1 EA: 0 < TEI: 000EA: 1 < N(S): 002 0: 0 < N(R): 003 P: 0 < 5 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 2 to (but not including) 3 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter < Protocol Discriminator: Q.931 (8) len=5 < Call Ref: len= 2 (reference 6321/0x18B1) (Originator) < Message type: RELEASE (77) -- Making new call for cr 6321 > [ 00 01 06 06 08 02 98 b1 5a 08 02 81 d1 ] > Informational frame: > SAPI: 00 C/R: 0 EA: 0 > TEI: 000EA: 1 > N(S): 003 0: 0 > N(R): 003 P: 0 > 9 bytes of data Stopping T_203 timer Starting T_200 timer -- Restarting T200 timer > Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 6321/0x18B1) (Terminator) > Message type: RELEASE COMPLETE (90) > [08 02 81 d1] > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) > Ext: 1 Cause: Invalid call reference value (81), class = Invalid message (e.g. parameter out of range) (5) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Restarting T203 timer INV-VOICESW01*CLI> < [ 00 01 01 08 ] < Supervisory frame: < SAPI: 00 C/R: 0 EA: 0 < TEI: 000EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 004 P/F: 0 < 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 2 to (but not including) 4 -- ACKing packet 3, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Restarting T203 timer INV-VOICESW01*CLI> Disconnected from Asterisk server [r...@inv-voicesw01 asterisk]# = The maximum call duration I have made so far is 3min Kind regards Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] Using asterisk as the recording server
> On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote: >> On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen >> wrote: >> >> > On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: >> > > On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen >> > > >wrote: >> > > >> > > > On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: >> > > > > On Sun, Sep 6, 2009 at 10:47 PM, Research >> >> > > > wrote: >> > > > > >> > > > > > Hello team; >> > > > > > While am aware and active user of astersk monitor function for >> > > > recording, i >> > > > > > would like to know if i can use asterisk as a pure recording >> > > > server(like >> > > > > > nice or witness) for some other PABX's extensions (both >> inbound, >> > > > outbound >> > > > > > and internal). >> > > > > > >> > > > > > Setup >> > > > > > PSTN---Legacy PABX(with analogy n digital extensions)--- >> > > > asterisk(record >> > > > > > Legacy PABX extensions.) >> > > > > > >> > > > > > Sam >> > > > > > >> > > > > > >> > > > > Is there any SIP or other VoIP in the mix? If so, you should >> take a >> > look >> > > > at >> > > > > OrecX. >> > > > > http://oreka.sourceforge.net (Open Source) >> > > > > They also have a paid version. >> > > > >> > > > Another method to do that is to make the Asterisk monitor output >> dummy >> > > > SIP calls rather than sound files. Oreka/Orex can listen to those. >> > > > >> > > > Looking for volunteers to test that: >> > > > >> > > > http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ >> > > > http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ >> > > > >> > > > >> > http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample >> > > > >> > > > This allows recording non-VoIP links, VoIP links where tapping is >> not >> > > > convinient, or more selective recording of VoIP calls. >> > > > >> > > >> > > Is this similar or the same as the portion of my post that you >> snipped? >> > >> > Different in many ways, which is why I snipped it. >> > >> > > >> > > "Sangoma RTP Tap will allow you to record TDM calls, again using >> OrecX >> > but >> > > minus the VoIP." >> > >> > (Actually: recorded calls are sent as RTP streams to the Orex/Oreka >> > server) >> > >> > This records outside of Asterisk. Thus it lacks information available >> in >> > Asterisk (who really called who). OTOH, it is Asterisk-specific. >> > >> > We actually considered implementing something similar to the Sangoma >> > interface in our driver but realised that doing it in Asterisk would >> > probably be more useful. The overheade seems reasonable. >> > >> > >> Sorry, I fail to see the difference besides Sangoma implemented it in >> their >> Wanpipe drivers and you are attempting copy their idea and do it in >> Asterisk. >> >> Your quote "This allows recording non-VoIP links, VoIP links where >> tapping >> is not convenient (edited to fix your spelling mistake), or more >> selective >> recording of VoIP calls." >> >> Isn't that more or less the same thing I said that you snipped, "Sangoma >> RTP >> Tap will allow you to record TDM calls, again using OrecX but minus the >> VoIP." > > And what if the call does not go through a TDM card? And ore > importantly: how can you tell who is the caller and who is the callee? > The rtp-tap interface basically tells you that channel X had a call at > time Y. > > If you control recording through the monitoring interface of Asterisk > you can start and stop the recording when you need it. You can also > provide better information aobut the call. But again, it means that this > is part of Asterisk, and I figure Sangoma has quite a few non-Asterisk > customers. > >> >> This isn't the biz list, nor the dev list. Snipping out the reference >> of >>
[asterisk-users] Using asterisk as the recording server
Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AddQueueMember with Agents.conf
Hello Team As you are all aware, digium has removed agentcallbacklogin as from 1.6. Is anyone knows any work around to have say 20seats (SIP Clients), 100 agents call center for which user will have to login to the queue dynamically from any extension and yet populate queue information with own's information instead of SIP or Local channels for reporting purpose I have tried both AddQueueMember(,Local/@context) and AddQueueMember(,agent/SIP/) in vain Please advice Thanks Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI Error and poor audio quality
-- >> I know it doesn't really sound very helpful to blame the entire server >> manufacturer, but some others might agree, brand spanking new and shiny >> might not be the best thing for Asterisk, especially these cards. > There's nothing wrong with brand spanking new and shiny, as long as it is > not > certain name brand manufacturers who find a need to 'distinguish' >themselves in the marketplace by making motherboards that aren't fully standards compliant. > I've had far fewer problems with Dell, for example. Yes it is DL580 from HP. I wanted something big for the type of load to be used but now im very convinced that it wasn't a best shot! The major problem I have been facing with both dell and HP, is the kernel panic!! This one however doesn't give the panic but HDLC and D-channels disconnection does not want to dis appear. Digium has advised me to downgrade to dahdi 2.1.0.4 libpri 1.4.10.1 and monitor the situation. Busy doing that Shall update ___ -- 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] DAHDI Error and poor audio quality
Hello Team
I have installed the new DL580 and used the new TE420B to add capacity on
our ivr. Before I put new E1s I decided to first move the old e1 from the
old system to this new one but it has errors which not only affect the
audio quality, but also cause the asterisk to refuse any call after
sometime even though the channels seems up and active {seems d-channel
fails}..
When processing calls, too much cripping and poor quality sound
Find attached log extract
Linux voicesw09 2.6.18-92.el5xen
Asterisk 1.4.25.1
Dahdi 2.2.0
Libpri 1.4.10
[r...@pbx09 src]# dahdi_hardware
pci::23:08.0 wct4xxp+ d161:0420 Wildcard TE420 (4th Gen)
pci::26:08.0 wct4xxp+ d161:0420 Wildcard TE420 (4th Gen)
[r...@pbx09 src]#
[Jul 18 07:23:32] WARNING[15968] chan_dahdi.c: No D-channels available!
Using Primary channel 233 as D-channel anyway!
[Jul 18 07:23:34] WARNING[15962] chan_dahdi.c: No D-channels available!
Using Primary channel 47 as D-channel anyway!
[Jul 18 07:23:34] WARNING[15963] chan_dahdi.c: No D-channels available!
Using Primary channel 78 as D-channel anyway!
[Jul 18 07:23:34] WARNING[15964] chan_dahdi.c: No D-channels available!
Using Primary channel 109 as D-channel anyway!
[Jul 18 07:24:01] NOTICE[15961] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 1
[Jul 18 07:27:59] NOTICE[15964] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 4
[Jul 18 07:30:58] NOTICE[15963] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 3
[Jul 18 07:39:07] NOTICE[15967] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 7
[Jul 18 07:41:08] NOTICE[15967] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 7
[Jul 18 07:41:33] NOTICE[15961] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 1
[Jul 18 07:42:44] NOTICE[15967] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 7
[Jul 18 07:46:10] NOTICE[15964] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 4
[Jul 18 07:53:24] NOTICE[15967] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 7
[Jul 18 07:54:43] NOTICE[15968] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 8
[Jul 18 07:55:41] NOTICE[15967] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 7
[Jul 18 07:56:04] NOTICE[15968] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 8
[Jul 18 07:59:04] NOTICE[15961] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 1
[Jul 18 07:59:58] NOTICE[15968] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 8
[Jul 18 08:02:38] NOTICE[15968] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 8
[Jul 18 08:05:31] NOTICE[15962] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 2
[Jul 18 08:07:18] NOTICE[15966] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 6
[Jul 18 08:09:38] NOTICE[15964] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 4
[Jul 18 08:10:00] NOTICE[15962] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 2
[Jul 18 08:12:15] NOTICE[15965] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 5
[Jul 18 08:13:00] NOTICE[15963] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 3
[Jul 18 08:13:16] NOTICE[15962] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 2
[Jul 18 08:13:18] NOTICE[15964] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 4
[Jul 18 08:13:18] NOTICE[15964] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 4
[Jul 18 08:13:50] NOTICE[15965] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 5
[Jul 18 08:14:36] NOTICE[15962] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 2
[Jul 18 08:14:58] NOTICE[15963] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 3
[Jul 18 08:17:41] NOTICE[15961] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 1
[Jul 18 08:19:15] NOTICE[15963] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 3
[Jul 18 08:19:34] NOTICE[15963] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 3
[Jul 18 08:20:48] NOTICE[15966] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 6
[Jul 18 08:20:53] NOTICE[15961] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 1
[Jul 18 08:20:57] NOTICE[15961] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 1
[Jul 18 08:29:03] NOTICE[26605] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 6
[Jul 18 08:31:12] NOTICE[26600] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-ch
Re: [asterisk-users] SIP CALL: RTP ENCRYPTION
I have recompiled asterisk-srtp with #./configure --without-ss7 and everythink works.. now testing srtp functionality. Sam > I have been trying to install asterisk-srtp from branches but i get the > following error. > >[CC] chan_alsa.c -> chan_alsa.o > >[LD] chan_alsa.o -> chan_alsa.so > >[CC] chan_bridge.c -> chan_bridge.o > >[LD] chan_bridge.o -> chan_bridge.so > >[CC] chan_dahdi.c -> chan_dahdi.o > > chan_dahdi.c: In function âss7_reset_linksetâ: > > chan_dahdi.c:11104: warning: passing argument 2 of âisup_grsâ makes > pointer from integer without a cast > > chan_dahdi.c:11104: error: too many arguments to function âisup_grsâ > > chan_dahdi.c: In function âss7_linksetâ: > > chan_dahdi.c:11461: warning: passing argument 2 of âisup_graâ makes > pointer from integer without a cast > > chan_dahdi.c:11461: warning: passing argument 4 of âisup_graâ makes > pointer from integer without a cast > > chan_dahdi.c:11667: warning: passing argument 2 of âisup_cgbaâ makes > pointer from integer without a cast > > chan_dahdi.c:11667: warning: passing argument 4 of âisup_cgbaâ makes > pointer from integer without a cast > > chan_dahdi.c:11667: error: too many arguments to function âisup_cgbaâ > > chan_dahdi.c:11677: warning: passing argument 2 of âisup_cguaâ makes > pointer from integer without a cast > > chan_dahdi.c:11677: warning: passing argument 4 of âisup_cguaâ makes > pointer from integer without a cast > > chan_dahdi.c:11677: error: too many arguments to function âisup_cguaâ > > chan_dahdi.c:11703: warning: passing argument 2 of âisup_blaâ makes > pointer from integer without a cast > > chan_dahdi.c:11703: error: too many arguments to function âisup_blaâ > > chan_dahdi.c:11728: warning: passing argument 2 of âisup_ubaâ makes > pointer from integer without a cast > > chan_dahdi.c:11728: error: too many arguments to function âisup_ubaâ > > chan_dahdi.c: In function âlinkset_addsigchanâ: > > chan_dahdi.c:15324: warning: passing argument 3 of âss7_add_linkâ makes > pointer from integer without a cast > > chan_dahdi.c:15324: error: too few arguments to function âss7_add_linkâ > > chan_dahdi.c:15326: warning: passing argument 3 of âss7_add_linkâ makes > pointer from integer without a cast > > chan_dahdi.c:15326: error: too few arguments to function âss7_add_linkâ > > chan_dahdi.c:15350: warning: implicit declaration of function > âss7_set_adjpcâ > > chan_dahdi.c: In function âhandle_ss7_block_cicâ: > > chan_dahdi.c:15449: warning: passing argument 2 of âisup_bloâ makes > pointer from integer without a cast > > chan_dahdi.c:15449: error: too many arguments to function âisup_bloâ > > chan_dahdi.c: In function âhandle_ss7_block_linksetâ: > > chan_dahdi.c:15505: warning: passing argument 2 of âisup_bloâ makes > pointer from integer without a cast > > chan_dahdi.c:15505: error: too many arguments to function âisup_bloâ > > chan_dahdi.c: In function âhandle_ss7_unblock_cicâ: > > chan_dahdi.c:15559: warning: passing argument 2 of âisup_ublâ makes > pointer from integer without a cast > > chan_dahdi.c:15559: error: too many arguments to function âisup_ublâ > > chan_dahdi.c: In function âhandle_ss7_unblock_linksetâ: > > chan_dahdi.c:15609: warning: passing argument 2 of âisup_ublâ makes > pointer from integer without a cast > > chan_dahdi.c:15609: error: too many arguments to function âisup_ublâ > > make[1]: *** [chan_dahdi.o] Error 1 > > make: *** [channels] Error 2 > > [r...@invpbx02 asterisk-srtp]# > > > someone with a good idea on how to resolve this? seems i can not install > any 1.6 version > > Sam >>> On Thu, May 28, 2009 at 02:00:15PM -0500, resea...@businesstz.com >>> wrote: Hello May i please know if asterisk is now supporting sip call encryption. It has been a requirement from one of my client to ensure that all conversation is well secured from any potential sniffers or inside hackers I have reviewed and shall soon try: http://www.voip-info.org/wiki/view/Asterisk+SRTP >>> >>> This technically isn't SIP encryption. It encrypts the RTP streams. >>> Though this is probably what you're really after. >>> >>> This still won't e.g. encrypt the dialed number. >>> >>> -- >>>Tzafrir Cohen >>> icq#16849755 jabber:tzafrir.co...@xorcom.com >>> +972-50-7952406 mailto:tzafrir.co...@xorcom.com >>> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir >>> >>> >>> >>> -- >> >> Thanks Tzafrir. yes, the aim is to encrypt the rtp streams. so any good >> idea!! So has any one done this in production? >> >> Kind regards >> Sam >> > > ___ -- 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] SIP CALL: RTP ENCRYPTION
I have been trying to install asterisk-srtp from branches but i get the following error. [CC] chan_alsa.c -> chan_alsa.o [LD] chan_alsa.o -> chan_alsa.so [CC] chan_bridge.c -> chan_bridge.o [LD] chan_bridge.o -> chan_bridge.so [CC] chan_dahdi.c -> chan_dahdi.o chan_dahdi.c: In function âss7_reset_linksetâ: chan_dahdi.c:11104: warning: passing argument 2 of âisup_grsâ makes pointer from integer without a cast chan_dahdi.c:11104: error: too many arguments to function âisup_grsâ chan_dahdi.c: In function âss7_linksetâ: chan_dahdi.c:11461: warning: passing argument 2 of âisup_graâ makes pointer from integer without a cast chan_dahdi.c:11461: warning: passing argument 4 of âisup_graâ makes pointer from integer without a cast chan_dahdi.c:11667: warning: passing argument 2 of âisup_cgbaâ makes pointer from integer without a cast chan_dahdi.c:11667: warning: passing argument 4 of âisup_cgbaâ makes pointer from integer without a cast chan_dahdi.c:11667: error: too many arguments to function âisup_cgbaâ chan_dahdi.c:11677: warning: passing argument 2 of âisup_cguaâ makes pointer from integer without a cast chan_dahdi.c:11677: warning: passing argument 4 of âisup_cguaâ makes pointer from integer without a cast chan_dahdi.c:11677: error: too many arguments to function âisup_cguaâ chan_dahdi.c:11703: warning: passing argument 2 of âisup_blaâ makes pointer from integer without a cast chan_dahdi.c:11703: error: too many arguments to function âisup_blaâ chan_dahdi.c:11728: warning: passing argument 2 of âisup_ubaâ makes pointer from integer without a cast chan_dahdi.c:11728: error: too many arguments to function âisup_ubaâ chan_dahdi.c: In function âlinkset_addsigchanâ: chan_dahdi.c:15324: warning: passing argument 3 of âss7_add_linkâ makes pointer from integer without a cast chan_dahdi.c:15324: error: too few arguments to function âss7_add_linkâ chan_dahdi.c:15326: warning: passing argument 3 of âss7_add_linkâ makes pointer from integer without a cast chan_dahdi.c:15326: error: too few arguments to function âss7_add_linkâ chan_dahdi.c:15350: warning: implicit declaration of function âss7_set_adjpcâ chan_dahdi.c: In function âhandle_ss7_block_cicâ: chan_dahdi.c:15449: warning: passing argument 2 of âisup_bloâ makes pointer from integer without a cast chan_dahdi.c:15449: error: too many arguments to function âisup_bloâ chan_dahdi.c: In function âhandle_ss7_block_linksetâ: chan_dahdi.c:15505: warning: passing argument 2 of âisup_bloâ makes pointer from integer without a cast chan_dahdi.c:15505: error: too many arguments to function âisup_bloâ chan_dahdi.c: In function âhandle_ss7_unblock_cicâ: chan_dahdi.c:15559: warning: passing argument 2 of âisup_ublâ makes pointer from integer without a cast chan_dahdi.c:15559: error: too many arguments to function âisup_ublâ chan_dahdi.c: In function âhandle_ss7_unblock_linksetâ: chan_dahdi.c:15609: warning: passing argument 2 of âisup_ublâ makes pointer from integer without a cast chan_dahdi.c:15609: error: too many arguments to function âisup_ublâ make[1]: *** [chan_dahdi.o] Error 1 make: *** [channels] Error 2 [r...@invpbx02 asterisk-srtp]# someone with a good idea on how to resolve this? seems i can not install any 1.6 version Sam >> On Thu, May 28, 2009 at 02:00:15PM -0500, resea...@businesstz.com wrote: >>> Hello >>> >>> May i please know if asterisk is now supporting sip call encryption. >>> It >>> has been a requirement from one of my client to ensure that all >>> conversation is well secured from any potential sniffers or inside >>> hackers >>> >>> I have reviewed and shall soon try: >>> http://www.voip-info.org/wiki/view/Asterisk+SRTP >> >> This technically isn't SIP encryption. It encrypts the RTP streams. >> Though this is probably what you're really after. >> >> This still won't e.g. encrypt the dialed number. >> >> -- >>Tzafrir Cohen >> icq#16849755 jabber:tzafrir.co...@xorcom.com >> +972-50-7952406 mailto:tzafrir.co...@xorcom.com >> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir >> >> >> >> -- > > Thanks Tzafrir. yes, the aim is to encrypt the rtp streams. so any good > idea!! So has any one done this in production? > > Kind regards > Sam > ___ -- 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] SIP CALL: RTP ENCRYPTION
> On Thu, May 28, 2009 at 02:00:15PM -0500, resea...@businesstz.com wrote: >> Hello >> >> May i please know if asterisk is now supporting sip call encryption. It >> has been a requirement from one of my client to ensure that all >> conversation is well secured from any potential sniffers or inside >> hackers >> >> I have reviewed and shall soon try: >> http://www.voip-info.org/wiki/view/Asterisk+SRTP > > This technically isn't SIP encryption. It encrypts the RTP streams. > Though this is probably what you're really after. > > This still won't e.g. encrypt the dialed number. > > -- >Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > > > -- Thanks Tzafrir. yes, the aim is to encrypt the rtp streams. so any good idea!! So has any one done this in production? Kind regards Sam ___ -- 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] SIP CALL ENCRYPTION
Hello May i please know if asterisk is now supporting sip call encryption. It has been a requirement from one of my client to ensure that all conversation is well secured from any potential sniffers or inside hackers I have reviewed and shall soon try: http://www.voip-info.org/wiki/view/Asterisk+SRTP Please help or suggest any solution that you feel may help Kind regards Sam ___ -- 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] SIP CALL ENCRYPTION
Hello May i please know if asterisk is now supporting sip call encryption. It has been a requirement from one of my client to ensure that all conversation is well secured from any potential sniffers or inside hackers Please help or suggest any solution that you feel may help Kind regards Sam ___ -- 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] CALL SETUP TIME
Greetings List Im interested to know how long the setup time is for a particular call on asterisk. Is there any defined parameter that i can use to real this behavior? SETUP TIME = TIME BEFORE THE B-PART START RINGING Thank you in advance Sam ___ -- 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] Need some information on SS7 parameters
Thanks Matt I will speak to voda to know exactly parameter name and let your know soon Regards Sam > resea...@businesstz.com wrote: >> Can someone assist me on this please? >> >> >>> Hello List >>> >>> I am setting up a small demo site using SS7 and one of the requirement >>> is >>> to be able to unhide the numbers and locate exact location of the >>> caller >>> (BTS ID). Vodafone uses Nokia-Siemens switch and has confirmed that the >>> parameters will be sent to the us. >>> >>> I just want to know how do read those information from the dialplan to >>> be >>> able to present them to the Agent > > It depends on what parameter this information is encoded inside. > > If you can find out the name of the parameter, we could probably answer > your question. > > The likely answer is that we probably do not decode/expose this > parameter to the dialplan at this time, but adding and exposing > parameters is not a very hard thing to do. > > Matthew Fredrickson > Digium, Inc. > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some information on SS7 parameters
Can someone assist me on this please? > Hello List > > I am setting up a small demo site using SS7 and one of the requirement is > to be able to unhide the numbers and locate exact location of the caller > (BTS ID). Vodafone uses Nokia-Siemens switch and has confirmed that the > parameters will be sent to the us. > > I just want to know how do read those information from the dialplan to be > able to present them to the Agent > > Thanks > Samwel Muro > ___ -- 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] Need some information on SS7 parameters
Hello List I am setting up a small demo site using SS7 and one of the requirement is to be able to unhide the numbers and locate exact location of the caller (BTS ID). Vodafone uses Nokia-Siemens switch and has confirmed that the parameters will be sent to the us. I just want to know how do read those information from the dialplan to be able to present them to the Agent Thanks Samwel Muro ___ -- 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] Ring/Off-hook in strange state 6 channel X
Versions - Asterisk 1.4.22 - DAHDI Linux 2.0.0 - DAHDI Tools 2.0.0 - Libpri 1.4.7 - Addons 1.4.7 Here is chan_dahdi.conf ; ; DAHDI telephony interface [trunkgroups] [channels] context=from-pstn switchtype=national signalling=fxo_ks rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=no threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes immediate=no busydetect=no callprogress=no answeronpolarityswitch=yes hanguponpolarityswitch=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 #include dahdi-channels.conf File: dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Mon Nov 24 16:19:00 2008 -- do not hand edit ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/asterisk/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER) ;;; line="3 WCTDM/0/2 FXSKS (EC: MG2)" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 3 callerid= group= context=default ; Span 2: XBUS-00/XPD-00 "Xorcom XPD #00/00: FXO" ;;; line="5 XPP_FXO/00/00/0" signalling=fxs_ks callerid=asreceived cidsignalling=v23 cidstart=polarity callerid=asreceived callwaiting=no group=0 context=from-pstn channel => 5-12 > Greetings List > > I have connected my asterisk box with x100 2xfxo and xorcom 8xfxo and all > of them give me the error "Ring/Off-hook in strange state 6". > > Whenever the caller hangup, the call continue to execute until it hits the > hard coded hangup. I changed chan_dadhi busydetect=no and callprogress=no > but problem still persist. I also tried to use different PABX in vain. GSM > modem (FUSION100) also produces no useful result > > Please help > > Sam Muro > ___ -- 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] Ring/Off-hook in strange state 6 channel X
Greetings List I have connected my asterisk box with x100 2xfxo and xorcom 8xfxo and all of them give me the error "Ring/Off-hook in strange state 6". Whenever the caller hangup, the call continue to execute until it hits the hard coded hangup. I changed chan_dadhi busydetect=no and callprogress=no but problem still persist. I also tried to use different PABX in vain. GSM modem (FUSION100) also produces no useful result Please help Sam Muro ___ -- 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 as an IVR
Oh Edward
You are my Hero... Simple but perfect. Option II is ideal but as you know
this is Asterisk/*/everything..
Thanks to list
Kill
>> Can someone assist to unfold the secret on how to atleast to a count on
>> particular branch, say, if 2 is chosen, then we start count from the
>> time
>> the choice is made to the time the caller hangup or choice another
>> option
>>
>> i.e.
>> exten => s,1,Answer()
>> exten => s,n,Background(PLEASE ENTER YOU OPTION)
>>
>> exten => s,n,XXX ; //IS IT POSSIBLE TO START A STOPWATCH (COUNTER) HERE
>> exten => s,n,WaitExten(10)
>> exten => s,n,Goto(s,1)
>>
>> exten => 1,1,Answer()
>> exten => 1,n,AGI(POPULATE PREVIOUS BRANCH DATA INTO THE DB)
>> exten => 1,n,XXX ;(RE)START A COUNTER HERE
>> exten => 1,n,PLAYBACK(OPTION1 SELECTED);(RE)START A COUNTER HERE
>> exten => 1,n,Hangup
>>
>> exten => 2,1,Answer()
>> exten => 2,n,AGI(POPULATE PREVIOUS BRANCH DATA INTO THE DB)
>> exten => 2,n,XXX ;(RE)START A COUNTER HERE
>> exten => 2,n,PLAYBACK(OPTION2 SELECTED);(RE)START A COUNTER HERE
>> exten => 2,n,Hangup
>
> You could "start" your stopwatch with
>
> exten = s,n,set(STOPWATCH=${EPOCH})
>
> instead of your extraneous answer()s and then in your "h" extension, stop
> the stopwatch with:
>
> exten = h,n,set(STOPWATCH=$[${EPOCH} -
> ${STOPWATCH}])
>
> You could also use resetcdr(w) at the start of each option. This will
> create a new CDR with the time spent on the previous option at each step
> in your dialplan. By setting the option number in a CDR variable after
> each CDR is written, the time spent in each option can be identified.
>
> Thanks in advance,
>
> Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
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Re: [asterisk-users] Asterisk as an IVR
Greetings
Can someone assist to unfold the secret on how to atleast to a count on
particular branch, say, if 2 is chosen, then we start count from the time
the choice is made to the time the caller hangup or choice another option
i.e.
exten => s,1,Answer()
exten => s,n,Background(PLEASE ENTER YOU OPTION)
exten => s,n,XXX ; //IS IT POSSIBLE TO START A STOPWATCH (COUNTER) HERE
exten => s,n,WaitExten(10)
exten => s,n,Goto(s,1)
exten => 1,1,Answer()
exten => 1,n,AGI(POPULATE PREVIOUS BRANCH DATA INTO THE DB)
exten => 1,n,XXX ;(RE)START A COUNTER HERE
exten => 1,n,PLAYBACK(OPTION1 SELECTED);(RE)START A COUNTER HERE
exten => 1,n,Hangup
exten => 2,1,Answer()
exten => 2,n,AGI(POPULATE PREVIOUS BRANCH DATA INTO THE DB)
exten => 2,n,XXX ;(RE)START A COUNTER HERE
exten => 2,n,PLAYBACK(OPTION2 SELECTED);(RE)START A COUNTER HERE
exten => 2,n,Hangup
i believe we can set something very powerful here
Kili
> Thanks Anselm
>
> Its true that is a lot of calls but i have a separate mysql database on
> different server (HP DL580G5 with 16cores). what am currently doing is
> capturing the information right after selection and insert that record
> into mySql.
>
>
> [macro-capture-input]
> ;
> ;
> ; Macro that feeds data into mysql through perl script:
> ; ${ARG1} - MSISDN
> ; ${ARG2} - src
> ; ${ARG3} - MainMenu Application
> ; ${ARG4} - Channel
> ; ${ARG5} - calldatetime
> ; ${ARG6} - Sub Menu Application
> ;
> ;
> exten => s,1,System(/var/lib/asterisk/agi-bin/capture.pl ${ARG1} ${ARG2}
> ${ARG3} ${ARG4} ${ARG5} ${ARG6})
>
> [Data-Services-Options]
> ; This menu is aimed to provide user with info about data services
> offered by Vodacom, including
> ; 1 - SUBMenu 1
> ; 2 - SUBMenu 2
> ; 3 - SUBMenu 3
> ; 4 - SUBMenu 4
> ;
> ;
> ;SUBMENU 1
> ;
> exten =>
> 1,1,Macro(capture-input,"${MSISDN}","${OPT}","APPLICATION1","${CHANNEL}","now()","SUBMENU1")
> exten => 1,n,Background(IVR/(1110) MENU 1)
>
> ;SUBMENU 2
> exten => 2,1,Macro(sendsms,${MSISDN},1,${LANGUAGE})
> exten =>
> 2,n,Macro(ivrcdr,"${MSISDN}","${OPT}","APPLICATION2","${CHANNEL}","now()","SUBMENU2")
> exten => 2,n,Background(IVR/(1120) MENU 2)
> .
> .
> .
>
>
> what i will also want to capture is how long a caller took to listen to
> say SUBMENU1
>
> It should be noted that CDR doesnot capture such detailed info (Tzafrir)
>
> Regards
> Kili
>>
>> On Sat, Jun 28, 2008 at 03:37:56PM +0200, Anselm Martin Hoffmeister
>> wrote:
>>> Am Samstag, den 28.06.2008, 08:15 -0500 schrieb
>>> [EMAIL PROTECTED]:
>>> > Hi List
>>> >
>>> > I have build an IVR on Asterisk from 1.2 to now 1.4.18 and has
>>> already
>>> > processed more than 10million calls!
>>> >
>>> > I have one big challenge which is reporting... it is the requirement
>>> to
>>> > have a web reporting module which should the following info based on
>>> > selected time frame
>>> > - Number of calls on specific branch- Done
>>> > - Number of calls to branch 1 that came from branch 2 (this should
>>> be
>>> > flexible)
>>> > - talktime on specified branch (say how long caller listened to
>>> option
>>> 1
>>> > before choosing option 2 or hangup)
>>> >
>>> > On IVR, it is so important to understand how many callers select a
>>> > specific branch and how long they spent on that branch. CDR stats can
>>> not
>>> > provide these type of information and on trying freepbx, still can
>>> not
>>> go
>>> > so detailed
>>>
>>> Dear Kili,
>>>
>>> in my opinion this is a good application for Database backends. You
>>> could, for example, write entries to a DB whenever someone presses a
>>> key
>>> (or is re-routed in the dialplan, which comes to a similar scheme). In
>>> data mining time some SQL logic can produce nearly any data you want,
>>> provided the input data is there.
>>>
>>> Millions of calls sounds a lot though, so be sure to have a reasonable
>>> database backend: The asterisk included one might be a bit on the small
>>> side here.
>>>
>>> This is just an idea, I did not implement anything the like (yet).
>>
>> Asterisk already has this separate database backend: CDR.
>>
>> --
>>Tzafrir Cohen
>> icq#16849755 jabber:[EMAIL PROTECTED]
>> +972-50-7952406 mailto:[EMAIL PROTECTED]
>> http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
>>
>
>
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Re: [asterisk-users] Asterisk as an IVR
Thanks Anselm
Its true that is a lot of calls but i have a separate mysql database on
different server (HP DL580G5 with 16cores). what am currently doing is
capturing the information right after selection and insert that record
into mySql.
[macro-capture-input]
;
;
; Macro that feeds data into mysql through perl script:
; ${ARG1} - MSISDN
; ${ARG2} - src
; ${ARG3} - MainMenu Application
; ${ARG4} - Channel
; ${ARG5} - calldatetime
; ${ARG6} - Sub Menu Application
;
;
exten => s,1,System(/var/lib/asterisk/agi-bin/capture.pl ${ARG1} ${ARG2}
${ARG3} ${ARG4} ${ARG5} ${ARG6})
[Data-Services-Options]
; This menu is aimed to provide user with info about data services
offered by Vodacom, including
; 1 - SUBMenu 1
; 2 - SUBMenu 2
; 3 - SUBMenu 3
; 4 - SUBMenu 4
;
;
;SUBMENU 1
;
exten =>
1,1,Macro(capture-input,"${MSISDN}","${OPT}","APPLICATION1","${CHANNEL}","now()","SUBMENU1")
exten => 1,n,Background(IVR/(1110) MENU 1)
;SUBMENU 2
exten => 2,1,Macro(sendsms,${MSISDN},1,${LANGUAGE})
exten =>
2,n,Macro(ivrcdr,"${MSISDN}","${OPT}","APPLICATION2","${CHANNEL}","now()","SUBMENU2")
exten => 2,n,Background(IVR/(1120) MENU 2)
.
.
.
what i will also want to capture is how long a caller took to listen to
say SUBMENU1
It should be noted that CDR doesnot capture such detailed info (Tzafrir)
Regards
Kili
>
> On Sat, Jun 28, 2008 at 03:37:56PM +0200, Anselm Martin Hoffmeister wrote:
>> Am Samstag, den 28.06.2008, 08:15 -0500 schrieb [EMAIL PROTECTED]:
>> > Hi List
>> >
>> > I have build an IVR on Asterisk from 1.2 to now 1.4.18 and has already
>> > processed more than 10million calls!
>> >
>> > I have one big challenge which is reporting... it is the requirement
>> to
>> > have a web reporting module which should the following info based on
>> > selected time frame
>> > - Number of calls on specific branch- Done
>> > - Number of calls to branch 1 that came from branch 2 (this should be
>> > flexible)
>> > - talktime on specified branch (say how long caller listened to option
>> 1
>> > before choosing option 2 or hangup)
>> >
>> > On IVR, it is so important to understand how many callers select a
>> > specific branch and how long they spent on that branch. CDR stats can
>> not
>> > provide these type of information and on trying freepbx, still can not
>> go
>> > so detailed
>>
>> Dear Kili,
>>
>> in my opinion this is a good application for Database backends. You
>> could, for example, write entries to a DB whenever someone presses a key
>> (or is re-routed in the dialplan, which comes to a similar scheme). In
>> data mining time some SQL logic can produce nearly any data you want,
>> provided the input data is there.
>>
>> Millions of calls sounds a lot though, so be sure to have a reasonable
>> database backend: The asterisk included one might be a bit on the small
>> side here.
>>
>> This is just an idea, I did not implement anything the like (yet).
>
> Asterisk already has this separate database backend: CDR.
>
> --
>Tzafrir Cohen
> icq#16849755 jabber:[EMAIL PROTECTED]
> +972-50-7952406 mailto:[EMAIL PROTECTED]
> http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
>
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[asterisk-users] Asterisk as an IVR
Hi List I have build an IVR on Asterisk from 1.2 to now 1.4.18 and has already processed more than 10million calls! I have one big challenge which is reporting... it is the requirement to have a web reporting module which should the following info based on selected time frame - Number of calls on specific branch- Done - Number of calls to branch 1 that came from branch 2 (this should be flexible) - talktime on specified branch (say how long caller listened to option 1 before choosing option 2 or hangup) On IVR, it is so important to understand how many callers select a specific branch and how long they spent on that branch. CDR stats can not provide these type of information and on trying freepbx, still can not go so detailed Is there anyone with similar project that is willing to share some information Regards Kili ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
