Re: [asterisk-users] Asterisk Drop call
Roberto Check your router if ALG or similar feature is enabled. Disable and test. Also, on SNGREP check if both parties are getting ACK correctly after RTP starts. *--* *Atenciosamente,* *Luciano Moreira**(85)99974-2750* *__Logic Telecom* *0800-085-7799 | (85)4042-7799 | **(11)4210-7799* Em ter., 22 de set. de 2020 às 13:35, Roberto < roberto.med...@gasparimsantos.com.br> escreveu: > Hello. > Thanks for the reply. > > Yes. In the traffic analyzed, the BYE is sent by the originator of the > call, but there is no "human" hangup, but the asterisk one. > BYE is sent, received and confirmed. > > I don't know how I could investigate the reason for this BYE. > > Em 21/09/2020 17:12, Dovid Bender escreveu: > > Is there anything in the Asterisk logs? Which side sends the BYE? Were you > able to capture the traffic with sngrep/wireshark to see if any side > stopped sending/getting RTP? What did the other side see? > > > On Mon, Sep 21, 2020 at 3:22 PM Roberto < > roberto.med...@gasparimsantos.com.br> wrote: > >> Hello >> I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a >> drop in call. It does not have a certain time, it is random. The audio >> is flowing normally and the call is dropped. >> Has anyone ever experienced this? >> >> My settings changed below: >> >> allowoverlap = no >> udpbindaddr = 0.0.0.0 >> tcpenable = no >> tcpbindaddr = 0.0.0.0 >> >> transport = udp, ws, wss >> >> srvlookup = yes >> >> directmedia = no >> >> rtcachefriends = yes >> >> externaddr = my ip address >> >> externhost = my domain address ; foo.dyndns.net; refreshed periodically >> externrefresh = 180 >> >>localnet = 172.31.40.21 / 255.255.240.0; AWS NETWORK >>localnet = 192.168.0.0 / 255.255.0.0; RFC 1918 addresses >>localnet = 10.0.0.0 / 255.0.0.0; Also RFC1918 >>localnet = 172.16.0.0 / 12; Another RFC1918 with CIDR notation >>localnet = 169.254.0.0 / 255.255.0.0; Zero conf local network >>localnet = 200.0.0.0 / 24 >>localnet = 191.0.0.0 / 24 >>localnet = 201.0.0.0 / 24 >>localnet = 177.0.0.0 / 24 >> >>localnet = 179.0.0.0 / 24 >> >> >> Thanks >> >> Roberto. >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> 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 -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk distribution for a Call Center
We use Vicidial for all size CallCenter. It's very powerful for multi server and/or multi site. We have vicidial from tiny callcenter one site with 5 agents to over 1000 Agents distributed in 20 cities working as just one callcenter. Info http://astguiclient.sourceforge.net/vicidial.html __ Luciano Moreira Logic Telecom LTDa Fortaleza, CE +55 (85) 4062-9150 +55 (85) 9701-2444 +1 360-717-1506 (USA) 2010/6/22 Tarek Sawah tareksa...@hotmail.com: i have been struggling with call center Customers for a couple of years now.. i have a call center with 40 agents using elastix.. and quality is related to the source of calls inbound or outbound... the problem with call centers they need Visual .. like Flash Operator panel and CDRs.. if you can go with simply raw asterisk .. without any additions.. will be the best for you .. write your own dial plans. Flash operator Panel is not a flawless work.. and adds more burden on the resources.. esp when it's open by 7-8 persons at once.. regarding the ACD ..it's all about PHP and Database .. you can build your own reports and so. or you can use a2billing to do the billing and ACD.. Elastix has a good billing (without a2billing) .. but i prefer a clean installation of asterisk and work around with database and PHP much better.. Good Luck! -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 (386) 492-9993 Date: Tue, 22 Jun 2010 15:21:18 -0300 From: aco1...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk distribution for a Call Center Dear all, I need to build a PBX based on Asterisk for a call center. I have worked with raw Asterisk but it's hard to work for big implementations think. Also I have worked with Trixbox CE for a small bussines and it was prette good, but I have not have many features like ACD. I know there is another version called Trixbox PRO -specially Call Center edition- that's not free but has got more features like ACD and billing. I've heart about AsteriskNow and I know it's free. What distribution/version do you recommend to me in order to implement a call center and taking into account I'm not an expert in programming from Asterisk CLI ??? Thanks a lot Alejandro -- _ -- 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 Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. Learn more. -- _ -- 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] No outbound with A2Billing
List members, When I dial to a PSTN number, the A2Billing script does all the tasks, until it shutdown without make the dailout by sip trunk set. Lasts outputs fro the a2billing.php debug are: a2billing.php|2: RESFINDRATE:: 0 a2billing.php|2: UPDATE cc_card SET inuse=inuse-1 WHERE username='5033845534' Sip trunk is registered and working. All setups in A2Billing db seams ok. There is any a2billing guru to help me? Below, is the complete script output. Thank you in advance. Luc Moreira __ Logic Telecom Fortaleza, Brasil --- -- Accepting AUTHENTICATED call from 201.49.16.125: requested format = g723, requested prefs = (), actual format = g729, host prefs = (g729|g723|gsm|ulaw), priority = mine -- Executing Answer(IAX2/1010-15, ) in new stack -- Executing Wait(IAX2/1010-15, 0) in new stack -- Executing DeadAGI(IAX2/1010-15, a2billing.php|2) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|2: IDCONFIG : 2 a2billing.php|2: a2billing.php|2: A2Billing AGI internal configuration: a2billing.php|2: Array a2billing.php|2: ( a2billing.php|2: [debug] = 3 a2billing.php|2: [answer_call] = 1 a2billing.php|2: [logger_enable] = 1 a2billing.php|2: [log_file] = /tmp/a2billing.log a2billing.php|2: [say_goodbye] = a2billing.php|2: [play_menulanguage] = a2billing.php|2: [force_language] = br a2billing.php|2: [len_cardnumber] = 10 a2billing.php|2: [len_aliasnumber] = 5 a2billing.php|2: [len_voucher] = 15 a2billing.php|2: [min_credit_2call] = 0 a2billing.php|2: [min_duration_2bill] = 20 a2billing.php|2: [notenoughcredit_cardnumber] = 1 a2billing.php|2: [notenoughcredit_assign_newcardnumber_cid] = 1 a2billing.php|2: [use_dnid] = 1 a2billing.php|2: [no_auth_dnid] = Array a2billing.php|2: ( a2billing.php|2: [0] = a2billing.php|2: ) a2billing.php|2: a2billing.php|2: [number_try] = 1 a2billing.php|2: [say_balance_after_auth] = a2billing.php|2: [say_balance_after_call] = a2billing.php|2: [say_rateinitial] = a2billing.php|2: [say_timetocall] = a2billing.php|2: [auto_setcallerid] = 1 a2billing.php|2: [force_callerid] = a2billing.php|2: [cid_sanitize] = a2billing.php|2: [cid_enable] = 1 a2billing.php|2: [cid_askpincode_ifnot_callerid] = 1 a2billing.php|2: [cid_auto_create_card] = 1 a2billing.php|2: [cid_auto_assign_card_to_cid] = 1 a2billing.php|2: [cid_auto_create_card_typepaid] = POSTPAY a2billing.php|2: [cid_auto_create_card_credit] = 0 a2billing.php|2: [cid_auto_create_card_credit_limit] = 100 a2billing.php|2: [cid_auto_create_card_tariffgroup] = 1 a2billing.php|2: [callerid_authentication_over_cardnumber] = a2billing.php|2: [sip_iax_friends] = a2billing.php|2: [sip_iax_pstn_direct_call_prefix] = 9 a2billing.php|2: [sip_iax_pstn_direct_call] = a2billing.php|2: [dialcommand_param] = |60|HL(%timeout%:61000:3,Ttr) a2billing.php|2: [dialcommand_param_sipiax_friend] = |60|HL(360:61000:3,Ttr) a2billing.php|2: [switchdialcommand] = 1 a2billing.php|2: [maxtime_tocall_negatif_free_route] = 3600 a2billing.php|2: [send_reminder] = 1 a2billing.php|2: [record_call] = a2billing.php|2: [monitor_formatfile] = gsm a2billing.php|2: [base_currency] = usd a2billing.php|2: [agi_force_currency] = usd a2billing.php|2: [currency_association] = Array a2billing.php|2: ( a2billing.php|2: [0] = usd:prepaid-dollar a2billing.php|2: [1] = mxn:pesos a2billing.php|2: [2] = eur:euro a2billing.php|2: [3] = all:credit a2billing.php|2: [4] = brl:credit a2billing.php|2: ) a2billing.php|2: a2billing.php|2: [file_conf_enter_destination] = prepaid-enter-dest a2billing.php|2: [file_conf_enter_menulang] = prepaid-menulang2 a2billing.php|2: [setlanguage_deprecate] = 1 a2billing.php|2: [currency_association_internal] = Array a2billing.php|2: ( a2billing.php|2: [usd] = prepaid-dollar a2billing.php|2: [mxn] = pesos a2billing.php|2: [eur] = euro a2billing.php|2: [all] = credit a2billing.php|2: [brl] = credit a2billing.php|2: ) a2billing.php|2: a2billing.php|2: ) a2billing.php|2: a2billing.php|2: AGI Request: a2billing.php|2: Array a2billing.php|2: ( a2billing.php|2: [agi_request] = a2billing.php a2billing.php|2: [agi_channel] = IAX2/1010-15 a2billing.php|2: [agi_language] = br a2billing.php|2: [agi_type] = IAX2 a2billing.php|2: [agi_uniqueid] = 1156436221.21 a2billing.php|2: [agi_callerid] = 1010 a2billing.php|2:
[asterisk-users] Comfort noise support incomplete in Asterisk (RFC 3389).
I trying to setup a outbound trunk with IPSmarx. It's working, but when I make a call, the ring dialtone stills ringing on my side, even after the other side picksup the phone. I got a NOTICE message from Asterisk that I hope you can help me: -- Called [EMAIL PROTECTED] -- SIP/ipsmarx-out-0995f270 is making progress passing it to IAX2/1010-14 -- SIP/ipsmarx-out-0995f270 is ringing -- SIP/ipsmarx-out-0995f270 is making progress passing it to IAX2/1010-14 Aug 16 15:39:21 NOTICE[16215]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.230 ipsmarx-out is my outbound route. I got two SIP passing process. So I listen 2 ringtone and when the second ringtone start with a delay I got this NOTICE from asterisk:Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.230. I googled this error but could find a fix to this bug. Thank you in advance. Luc Moreira __ Logic Telecom Fortaleza, Brasil +55 (85) 3263-0372 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A2Billing - destination
Caros, I installed the A2Billing - v1.2.2 with Asterisk 1.2.10. All works ok, but when I try callout got a message saying the number in not available. Can you help with a step-by-step to make a card autenticate and dial a number? Thank you Luc Moreira Mais VoIP -- Accepting AUTHENTICATED call from 192.168.0.103: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (gsm), priority = mine -- Executing Answer(IAX2/1003-7, ) in new stack -- Executing Wait(IAX2/1003-7, 2) in new stack -- Executing DeadAGI(IAX2/1003-7, a2billing.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php: line:58 - IDCONFIG : 1 a2billing.php: a2billing.php: line:67 - MODE : standard a2billing.php: a2billing.php: A2Billing AGI internal configuration: a2billing.php: Array a2billing.php: ( a2billing.php: [debug] = 1 a2billing.php: [answer_call] = 1 a2billing.php: [logger_enable] = 1 a2billing.php: [log_file] = /tmp/a2billing.log a2billing.php: [say_goodbye] = a2billing.php: [play_menulanguage] = a2billing.php: [force_language] = EN a2billing.php: [intro_prompt] = a2billing.php: [len_cardnumber] = 10 a2billing.php: [len_aliasnumber] = 15 a2billing.php: [len_voucher] = 15 a2billing.php: [min_credit_2call] = 0 a2billing.php: [min_duration_2bill] = 0 a2billing.php: [notenoughcredit_cardnumber] = 1 a2billing.php: [notenoughcredit_assign_newcardnumber_cid] = 1 a2billing.php: [use_dnid] = a2billing.php: [no_auth_dnid] = Array a2billing.php: ( a2billing.php: [0] = 2400 a2billing.php: [1] = 2300 a2billing.php: ) a2billing.php: a2billing.php: [number_try] = 3 a2billing.php: [say_balance_after_auth] = a2billing.php: [say_balance_after_call] = a2billing.php: [say_rateinitial] = a2billing.php: [say_timetocall] = 1 a2billing.php: [auto_setcallerid] = 1 a2billing.php: [force_callerid] = a2billing.php: [cid_sanitize] = a2billing.php: [cid_enable] = 1 a2billing.php: [cid_askpincode_ifnot_callerid] = 1 a2billing.php: [cid_auto_create_card] = 1 a2billing.php: [cid_auto_assign_card_to_cid] = 1 a2billing.php: [cid_auto_create_card_typepaid] = POSTPAY a2billing.php: [cid_auto_create_card_credit] = 5 a2billing.php: [cid_auto_create_card_credit_limit] = 1000 a2billing.php: [cid_auto_create_card_tariffgroup] = 6 a2billing.php: [callerid_authentication_over_cardnumber] = 1 a2billing.php: [sip_iax_friends] = 1 a2billing.php: [sip_iax_pstn_direct_call_prefix] = 9 a2billing.php: [sip_iax_pstn_direct_call] = 1 a2billing.php: [extracharge_did] = Array a2billing.php: ( a2billing.php: [0] = 091 a2billing.php: ) a2billing.php: a2billing.php: [extracharge_fee] = Array a2billing.php: ( a2billing.php: [0] = 0.25 a2billing.php: [1] = 0.5 a2billing.php: ) a2billing.php: a2billing.php: [dialcommand_param] = |30|HL(%timeout%:61000:3) a2billing.php: [dialcommand_param_sipiax_friend] = |30|HL(360:61000:3) a2billing.php: [switchdialcommand] = 1 a2billing.php: [maxtime_tocall_negatif_free_route] = 5400 a2billing.php: [send_reminder] = a2billing.php: [record_call] = a2billing.php: [monitor_formatfile] = gsm a2billing.php: [base_currency] = usd a2billing.php: [agi_force_currency] = a2billing.php: [currency_association] = Array a2billing.php: ( a2billing.php: [0] = usd:prepaid-dollar a2billing.php: [1] = mxn:pesos a2billing.php: [2] = eur:euro a2billing.php: [3] = all:credit a2billing.php: ) a2billing.php: a2billing.php: [file_conf_enter_destination] = prepaid-enter-dest a2billing.php: [file_conf_enter_menulang] = prepaid-menulang2 a2billing.php: [currency_association_internal] = Array a2billing.php: ( a2billing.php: [usd] = prepaid-dollar a2billing.php: [mxn] = pesos a2billing.php: [eur] = euro a2billing.php: [all] = credit a2billing.php: ) a2billing.php: a2billing.php: ) a2billing.php: a2billing.php: AGI Request: a2billing.php: Array a2billing.php: ( a2billing.php: [agi_request] = a2billing.php a2billing.php: [agi_channel] = IAX2/1003-7 a2billing.php: [agi_language] = en a2billing.php: [agi_type] = IAX2 a2billing.php: [agi_uniqueid] = 1154482698.196 a2billing.php: [agi_callerid] = 1003 a2billing.php: [agi_calleridname] = Luc - Logic Telecom a2billing.php: [agi_callingpres] = 1 a2billing.php: [agi_callingani2] = 0 a2billing.php: [agi_callington] = 0
[Asterisk-Users] Wierd bug with MD3200
Pessoal, Facing wierd bug on * using MD3200 modem. It was working ok, then after a boot bug started: The calls that came out thru the Zap channel, stopped work. The * gets the call from an IAX client and set it as active, even before the destination rings, and finaly when someone pickup de phone the call stays mute. any clues about this bug? Luc --- Luciano Moreira Sip Phone #: 1-747-661-7629 Phone #: +55 (85) 3263-0372 +55 (85) 9956-2956 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call stays mute
Pessoal, Facing wierd bug on * using MD3200 modem. It was working ok, then after a boot bug started: The calls that came out thru the Zap channel, stopped work. The * gets the call from an IAX client and set it as active, even before the destination rings, and finaly when someone pickup de phone the call stays mute. any clues about this bug? Luc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
