Re: [asterisk-users] Set qualify = yes on trunk can't do outgoing call

aab/230> Em sex, 15 de fev de 2019 às 20:40, basti escreveu: > Hello, asterisk think my local phone (extension 20) is absent. > > On 15.02.19 23:26, Rafael dos Santos Saraiva wrote: > > Hi > > > > When you set qualify to yes, the asterisk "test&qu

Re: [asterisk-users] Set qualify = yes on trunk can't do outgoing call

Hi When you set qualify to yes, the asterisk "test" the sip trunk with OPTIONS messages, if no receive responses from this messages, it consider the trunk offline. Possibly your sip provider dont accept (and dont reply) sip options requests. Rafael S. Saraiva Porto Alegre - RS | Mobile: (51)

Re: [asterisk-users] Asterisk non-root - selinux - astdb

ttp://br.linkedin.com/pub/rafael-saraiva/52/aab/230> Em seg, 3 de dez de 2018 às 05:51, Jean Aunis escreveu: > Hello, > > I haven't tried but this post probably gives a solution : > > https://bugzilla.redhat.com/show_bug.cgi?id=1342733 > > Regards > > Jean Aunis &g

[asterisk-users] Asterisk non-root - selinux - astdb

Hi I'm trying to use Asterisk running as non-root user and selinux enabled. Asterisk is running ok, but astdb not works. When i try to put in astdb, console shows this message: WARNING[1853]: db.c:350 ast_db_put: Couldn't execute statment: SQL logic error or missing database CentOS 7.5.1804

Re: [asterisk-users] Reject call from Asterisk dialplan

Hi I guess is not possible send specific SIP response from dialplan in Asterisk, but you can send ISDN hangupcauses. In this case, to reject the call you can use Hangup(21). To do this remotly, my suggestion is create a context that pickup the call and execute hangup with cause 21. [image: Sua

Re: [asterisk-users] Asterisk put call on hold when receive 183 Session Progress with media address 0.0.0.0 in SDP

200, so placing the call on hold when no media is available > sounds logic. > > Le 06/10/2017 à 03:56, Rafael dos Santos Saraiva a écrit : > > Hi > > > Is it a normal behavior of Asterisk put a call on hold when receive a > Session Progress with media address 0.0.0.0 in SD

[asterisk-users] Asterisk put call on hold when receive 183 Session Progress with media address 0.0.0.0 in SDP

Hi Is it a normal behavior of Asterisk put a call on hold when receive a Session Progress with media address 0.0.0.0 in SDP? I believe the call on hold should be initiate with a re-invite. Thanks -- Att, Rafael Saraiva -- _

Re: [asterisk-users] Turn on SIP debugging from DialPlan

Hi I don't know if works, but you can try this: System(tcpdump -nq -s 0 -i eth0 -w /tmp/sip.pcap port 5060 or udp portrange 1-2 &); Wait(1); Dial(SIP/${EXTEN}); System(pkill tcpdump); Hangup; Or whitout RTP:

[asterisk-users] PRI error: link goes down when make calls

Hi I have a PRI link with a Brazilian telco when i make a call from Asterisk to PRI link the call doesn't complete and the link goes down (RED alarm), after this returns to status OK. Incoming calls works, but whitout audio The following link has the log on call at the moment when the link

[asterisk-users] DAHDI Dynamic span as INT device identificator

Hi Is it possible assign an INT identification to dynamic device in DAHDI? Example: This is device: DYN/eth/eth1/04:74:A1:00:0A:AE/0 I want call this as span 1 I saw the assigned-spans.conf and aparently can be this. Thanks in advance. -- [image: Sua Foto] Rafael S.

Re: [asterisk-users] using dynamic DAHDI loop back

Hi Insert this on first line of chan_dahdi.conf: [channels] [image: Sua Foto] Rafael S. Saraiva Porto Alegre - RS | Mobile: (51) 8174-7956 2016-03-13 10:01 GMT-03:00

[asterisk-users] set framing on dynamic interface DAHDI

Hi I working with DAHDI Dynamic Interfaces using ethernet boards. I need set the framing to CCS, but the documentation of DAHDI not refer to it. My question is: there is a way to do this? *system.conf* dynamic=eth,enp0s8/00:00:00:00:00:01/0,31,0 echocanceller=mg2,1-15,17-31 bchan=1-15,17-31

Re: [asterisk-users] Manipulating of a dialed sequence

Hi Try this: [pbx] exten => _1X.,1,Answer() same => n,Set(VAR=012345*543210) same => n,Set(VAR1=${CUT(VAR,*,1)});012345 same => n,Set(VAR1=${CUT(VAR,*,2)});543210 [image: Sua Foto] Rafael S. SaraivaPorto Alegre - RS | Mobile: (51) 8174-7956

Re: [asterisk-users] Busy level in Asterisk 11

...@gmail.comRafael S. SaraivaPorto Alegre - RS | Mobile: (51) 8174-7956 http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 https://plus.google.com/u/0/+RafaelSaraivaRS 2015-08-12 9:41 GMT-03:00 Joshua Colp jc...@digium.com: On Wed, Aug 12, 2015, at 09:34 AM, Rafael dos Santos Saraiva wrote: Hi Kia ora

[asterisk-users] Busy level in Asterisk 11

Hi I need to set the number of incoming calls to one, but the outgoing calls should be unlimited. I think the busylevel parameter is for it(incoming calls), but not works. My config is: cat sip.conf [general] [template](!) qualify=yes cc_agent_policy=generic cc_monitor_policy=generic

[asterisk-users] Update of dialed number on sip phones

Hi I have a dialplan that search a phone from dialed code, i.e: mysql table: code:1234 dest: +555133449966 query in odbc function: SELECT dest FROM my_table WHERE code = '${ARG1}' dialplan: exten = _#7,1,Set(DESTNO=${ODBC_query_dest_in_table(${EXTEN:2})}) same =

Re: [asterisk-users] Calling multiple phones at ones

Hi Ivan Using the following extensions: SIP/100 SIP/101 SIP/102 Example 1: [default] exten = _X.,1,Dial(SIP/100SIP/101SIP/102,30,tT) same = n,hangup Example 2: exten = _X.,1,Dial(SIP/100,10,tT) exten = _X.,2,Dial(SIP/101,10,tT) exten = _X.,3,Dial(SIP/102,10,tT) same = n,hangup This is the

Re: [asterisk-users] AEL keyword IfTime with variable on time range

) { NoOp(Boa noite); Playback(beep); } Thank's [image: Sua Foto] rafaels...@gmail.comRafael S. SaraivaPorto Alegre - RS | Mobile: (51) 8174-7956 http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 https://plus.google.com/u/0/+RafaelSaraivaRS 2015-05-12 14:39 GMT-03:00 Rafael dos Santos Saraiva

Re: [asterisk-users] AEL keyword IfTime with variable on time range

should try it and find out if it works. If it does, let us know. Regards; John *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael dos Santos Saraiva *Sent:* Tuesday, May 12, 2015 11:58 AM *To:* Asterisk Users Mailing

[asterisk-users] AEL keyword IfTime with variable on time range

Hi It's possible using a variable in the iftime keyword argument? E.g: context text { s = { timerange = '06:00-12:00|*|*|*'; ifTime(${timerange} { Playback(ivr/goodbye); } } } thanks [image: Sua Foto] rafaels...@gmail.comRafael S. SaraivaPorto Alegre - RS | Mobile: (51)

[asterisk-users] Asterisk 13 stable?

Hi The Asterisk 13 is already stable for production environment? thank's [image: Sua Foto] rafaels...@gmail.comRafael S. SaraivaPorto Alegre - RS | Mobile: (51) 8174-7956 http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 https://plus.google.com/u/0/+RafaelSaraivaRS --

Re: [asterisk-users] audio gain in SIP channel

Hi To using VOLUME function the syntax is: Set(VOLUME(rx)=+n) Set(VOLUME(rx)=-n) Set(VOLUME(tx)=+n) Set(VOLUME(tx)=-n) I think is not possible retrieve the value of the channel. Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 2014-07-24 7:52 GMT-03:00

Re: [asterisk-users] audio gain in SIP channel

I dont using these functions (AGC/ DENOISE). My suggestion... try invert the priorities: Set(DENOISE(tx)=on) Set(DENOISE(rx)=on) Set(AGC(rx)=) Set(AGC(rx)=) And try higher values.. is more easy the perception if the values are larger than default. Att, *Rafael dos Santos Saraiva* http

Re: [asterisk-users] Call Identifier Logging

(Motif/google/+${EXTEN:3}@voice.google.com,,r); hangup; } And worked perfectly. It would be interesting, the developer team add a variable to channel with this data. Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 2014-07-21 18:59 GMT

Re: [asterisk-users] Call Identifier Logging

Try this: CDR(userfield) = ${SHELL(asterisk -rx core show channel ${CHANNEL} | grep Call Identifer | cut -d: -f2 | cut -d[ -f2 | cut -d] -f1 | cut -d\n -f1):0:-1}; Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 2014-07-22 15:08 GMT-03:00 Steven Wheeler

Re: [asterisk-users] Call Identifier Logging

Really, a dialplan function would be best. I too don't like of an idea of using a external process to get internal variables, but when necessary... :( Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 2014-07-22 16:29 GMT-03:00 Steve Edwards asterisk

Re: [asterisk-users] CDR(dst) not set in AEL macro

(aparently, the return in macro no works). I will go try with others versions and report the status. Thank's. Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 2014-07-12 7:33 GMT-03:00 Johan Wilfer li...@jttech.se: 2014-07-11 15:38, Rafael dos Santos Saraiva

Re: [asterisk-users] How to log caller IP address in the CDR?

Hi Set(CDR(userfield)=${SIPPEER(${CALLERID(num),ip)}) If caller is SIP peer. Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 2014-07-14 14:10 GMT-03:00 Rafael rrich...@gmail.com: can you please tell me exactly which file to edit please

[asterisk-users] CDR(dst) not set in AEL macro

} = 1) { Set(__PICKUPMARK=${destno}); if(${ODBC_verify_user(${CALLERID(num)})} 0) { t = tT; } else { t = t; } } else { t = T; } Dial(${dialstring}/${destno},30,${t}); return; } Thank's. Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230

[asterisk-users] CDR(dst) in AEL macro

} = 1) { Set(__PICKUPMARK=${destno}); if(${ODBC_verify_user(${CALLERID(num)})} 0) { t = tT; } else { t = t; } } else { t = T; } Dial(${dialstring}/${destno},30,${t}); return; } Thank's. Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230

Re: [asterisk-users] Sippeers realtime with minimum table

Hi Joshua I've tried to create a view in a database, but Asterisk requires updatable fields in sippeers table (view), I cannot edit the structure of the tables. I chose for create a php script to read database and create the sip.conf file. Thank's Att, *Rafael dos Santos Saraiva* http

[asterisk-users] Sippeers realtime with minimum table

Hi there It's possible configure realtime mysql in Asterisk with a non standard sippeers table? I need using a sippeers table from other system (non Asterisk). This table has a minimal configuration. Thank's Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230

Re: [asterisk-users] func_odbc

COUNT(*) AS count FROM blacklist WHERE (calleridnum='${ARG1}' OR calleridnum = NULL) AND (dest='${ARG2}' OR dest = NULL) Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 2014-04-03 20:32 GMT-03:00 Bryant Zimmerman brya...@zktech.com: Hi All Anyone know how

[asterisk-users] Function REGEX

to works this function. Thank's Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Function REGEX

All working fine. Thank you for your help. Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 2014-03-31 12:29 GMT-03:00 Eric Wieling ewiel...@nyigc.com: Here is an example from one of my production dialplans same = n,ExecIf(${REGEX(^1205|^1256|^1850

[asterisk-users] Verbose only one context

Hi It's possible in Asterisk 1.8 enable verbose only in one context or extension? thanks Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] Reverse Charging Indication MFCR2

Hi It's possible verify the Reverse Charging Indication on mfcr2 link directly con dialplan? Thank's Att, *Rafael dos Santos Saraiva* Tel: (51) 8174-7956 *Digium Certified Asterisk Administrator (dCCA)* http://www.astdocs.com | http://br.linkedin.com/pub/rafael-saraiva/52/aab/230

Re: [asterisk-users] Voicemail variables on email subject

I noticed that the problem occurs when I use the variables ${VM_DUR} and ${VM_CALLERID}. Only the subject of the message, if the body is not the problem. Using UTF or utf the same problem occurs. Att, *Rafael dos Santos Saraiva* Tel: (51) 8174-7956 *Digium Certified Asterisk Administrator (dCCA

[asterisk-users] Voicemail variables on email subject

- Rafael 1570|16 Thank's Att, *Rafael dos Santos Saraiva* Tel: (51) 8174-7956 *Digium Certified Asterisk Administrator (dCCA)* http://www.astdocs.com | http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 -- _ -- Bandwidth

Re: [asterisk-users] Voicemail variables on email subject

I tried with utf-8, iso8859-1 and us-ascii. I used the Sendmail client, but now testing with mailcmd=cat /tmp/voicemail.txt The version of Asterisk is 1.8.22.0. Att, *Rafael dos Santos Saraiva* Tel: (51) 8174-7956 *Digium Certified Asterisk Administrator (dCCA)* http://www.astdocs.com | http

Re: [asterisk-users] Voicemail variables on email subject

When sending by SendMail the problem is the same in any email client. Att, *Rafael dos Santos Saraiva* Tel: (51) 8174-7956 *Digium Certified Asterisk Administrator (dCCA)* http://www.astdocs.com | http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 2013/7/25 jg webaccou...@jgoettgens.de

[asterisk-users] Performance Asterisk large installation on Vmware/Xen

with 2 agents, without call recording. It is best to use XEN or VMware? Which best version of Asterisk for this scenario? Thank you. Att, *Rafael dos Santos Saraiva* Tel: (51) 8174-7956 | (51) 3205-1504 http://www.astdocs.com | http://br.linkedin.com/pub/rafael-saraiva/52/aab/230

[asterisk-users] Extensions mask as variable?

Hi, How to define a extension mask as global variable in Ast 1.8? For example: [globals] MYVARIABLE = _15[7-9]X I tried this way but it did not work. Thanks Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Asterisk + Google Voice

Hi Is possible make calls from Asterisk with Google Voice? The settings are done in jabber.conf and gtalk.conf? I was able to receive calls from Gtalk on my Asterisk, but I would also like to generate calls to the PSTN via Google. Thank's Rafael Saraiva --

Re: [asterisk-users] Asterisk + Google Voice

Perfectly works!! Thanks Rafael Saraiva 2012/7/28 Matthew Jordan mjor...@digium.com - Original Message - From: Rafael dos Santos Saraiva rafaels...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, July 28

Re: [asterisk-users] Block Collect Calls on ISDN trunk

Richard I tried this, but it did not work. What can be the problem? [PABX] exten = _x.,1,Proceeding() same = n,GotoIf($[${CHANNEL(reversecharge)} =-1]?allow:block) same = n(allow),Dial(SIP/1584,30,tT)) same = n(block),Hangup() Att, Rafael Saraiva 2012/2/15 Richard Mudgett

Re: [asterisk-users] Block Collect Calls on ISDN trunk

Of *Rafael dos Santos Saraiva *Sent:* Friday, February 17, 2012 10:26 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Block Collect Calls on ISDN trunk ** ** Richard ** ** ** ** I tried this, but it did not work. What can be the problem

Re: [asterisk-users] Block Collect Calls on ISDN trunk

-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael dos Santos Saraiva *Sent:* Friday, February 17, 2012 11:07 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Block Collect Calls on ISDN trunk

Re: [asterisk-users] Block Collect Calls on ISDN trunk

Of *Rafael dos Santos Saraiva *Sent:* Friday, February 17, 2012 11:21 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Block Collect Calls on ISDN trunk ** ** The value is always -1. I must enable something in chan_dahdi to pass the correct

Re: [asterisk-users] Block Collect Calls on ISDN trunk

Reversecharge not appear in debug. I'm in Brazil, the signaling is different here? Att, Rafael Saraiva 2012/2/17 Richard Mudgett rmudg...@digium.com The value is always -1. I must enable something in chan_dahdi to pass the correct value? ++ [PABX]

[asterisk-users] Block Collect Calls on ISDN trunk

How to block collect calls on ISDN trunk? Thank's Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Block Collect Calls on ISDN trunk

Richard Can you give me an example of how to use this function? Att, Rafael Saraiva 2012/2/15 Richard Mudgett rmudg...@digium.com How to block collect calls on ISDN trunk? You need Asterisk v1.8 or later and check the value of CHANNEL(reversecharge) in your dialplan.

[asterisk-users] calleridname presentation Asterisk = Siemens

Hi I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i can't show the callerid name in the way Asterisk == Siemens. I realized that Asterisk send calleridname in format namePresentationAllowedSimple to Siemens e Siemens send calleridname in format namePresentationAllowedExtended.

Re: [asterisk-users] calleridname presentation Asterisk = Siemens

Hi I change for first way in Asterisk 1.8: [teste] include=rota00 exten=1504,1,Set(CALLERID(name-charset)=unknown) exten=1504,2,Dial(DAHDI/g1/${EXTEN},60,tTwW) exten=1504,3,Hangup() But, in debug of the span show the simple form: 1 namePresentationAllowedSimple Context Specific [0 0x00] = 1

[asterisk-users] Conference feature

Hi How to create the conference feature in Asterisk? Thank's Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Conference feature

I am referring to 3-way conference Att, Rafael Saraiva 2011/6/26 Flavio Miranda flaviormira...@hotmail.com Very simple.. Just edit the meetme.conf in /etc/asterisk like this : [rooms] conf = 888 And then, in /etc/asterisk/ extensions.conf , put something like that: [conference]

[asterisk-users] calleridname presentation Asterisk == Siemens

Hi I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i can't show the callerid name in the way Asterisk == Siemens. I realized that Asterisk send calleridname in format namePresentationAllowedSimple to Siemens e Siemens send calleridname in format namePresentationAllowedExtended.

Re: [asterisk-users] DAHDI span timeing source

Hi The timing source is the clock of the system. When a equipment is 0, the other should be 1. The correct is: 0=slave, 1=master. The default for private systems is slave. Att, Rafael Saraiva 2011/5/27 satish patel satish...@hotmail.com Hi There, We have very old asterisk 1.2 running in

Re: [asterisk-users] DAHDI span timeing source

the you should user span=1,1,0 Check out http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html -- Sent from my iPhone On May 27, 2011, at 4:27 PM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: Hi The timing source

Re: [asterisk-users] Pridialplan/ prilocaldialplan

Hi I made a mistake. I was putting the lines pridialplan and prilocaldialplan after the line channel in chan_dahdi.conf. The Asterisk does not read the lines after channel. Thank's Att, Rafael Saraiva 2011/5/23 Rafael dos Santos Saraiva rafaels...@gmail.com did not work!! Bug in Asterisk

Re: [asterisk-users] Pridialplan/ prilocaldialplan

did not work!! Bug in Asterisk?? :( Rafael 2011/5/20 Захаров Антон ins...@mail.ru Yeap, I couldn't set Private TON too. Try to set all _prefix variables in chan_dahdi.conf and use dynamic prilocaldialplan. On 19.05.2011 21:30, Rafael dos Santos Saraiva wrote: Hi I change

Re: [asterisk-users] how to user SIP realtime option

Hi Trying exclude rtcachefriends from your sip.conf and include the field rtchachefriends in table sip_buddies. And exclude the field qualify from sip_buddies. Set YES in field rtcachefriends. Att, Rafael Saraiva 2011/5/21 virendra bhati virbh...@gmail.com Hi List, After read the link

Re: [asterisk-users] Pridialplan/ prilocaldialplan

2011/5/19 Захаров Антон ins...@mail.ru Hello. To apply this settings you should restart dahdi (dahdi restart in CLI). About influence you could read here: http://markmail.org/message/rpd2aewiu2soostz On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote: Hi I'm beginner in list. I have

[asterisk-users] Pridialplan/ prilocaldialplan

Hi I'm beginner in list. I have doubts about the options pridialplan and prilocaldiaplan in chan_dahdi.conf. I interconnect the Asterisk with a Siemens PBX, but i saw that the changes in the file do not take effect in debug of the span or calling/called number. How to use this options? In that