similar to: Reverse Charging Indication <> MFCR2

Hi I have a problem w/ voicemail, the subject message is corruption when used voicemail variables, e.g. : voicemail.conf emailsubject=${VM_MAILBOX}|${VM_MSGNUM}|${VM_CALLERID}|${VM_DUR} Return: Subject: =?utf-8?Q?1504|12|=22Teste_-_Rafael=22_=3C1570=3E|0=3A16?= Expected: Subject: 1504|12|"Teste - Rafael" <1570>|16 Thank's Att, *Rafael dos Santos Saraiva* Tel: (51)

Hi I would like the opinion of you and if anyone has a similar scenario. I have a project for installation of a Asterisk server in a client with about 400 extensions. My question is whether this scenario carry an Asterisk virtualized. Will be used only extensions and trunks sip sip, 1 queue with 2 agents, without call recording. It is best to use XEN or VMware? Which best version of Asterisk for

Sorry, I forget to tell I tried, but not works. *Context:* context ivr_temp2 { s => { Proceeding(); str_time_01 = '06:00-12:00|*|*|*'; // Manh? ifTime (${str_time_01}) { Playback(ura/bom_dia); } } } The error is showed on "ael reload". *Console errors:* rs0000sr304*CLI> ael reload Command 'ael reload' failed.

You should try it and find out if it works. If it does, let us know. Regards; John From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rafael dos Santos Saraiva Sent: Tuesday, May 12, 2015 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [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] <rafaelsnsa at gmail.com>Rafael S. SaraivaPorto Alegre - RS | Mobile: (51) 8174-7956 <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>

Hi The Asterisk 13 is already stable for production environment? thank's [image: Sua Foto] <rafaelsnsa at gmail.com>Rafael 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> -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 call-limit=2 busylevel=1 callcounter=yes subscribecontext = hint allowsubscribe=yes [100](template)

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> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140326/4ed97cc9/attachment.html>

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> -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I'm using a macro to dial in a AEL dialplan. The problem is the macro do not set the field CDR(dst), showing only ~~s~~. I tried various configurations, but without solutions. This is the macro: macro dial-out(destno,dialstring,route_descr,interno) { __TRANSFER_CONTEXT=ipbx; if(${interno} = 1) { Set(__PICKUPMARK=${destno}); if(${ODBC_verify_user(${CALLERID(num)})} > 0) { t = tT; }

Dear Steve and *.* e1r2 developers and users, now MFCR2 is successfully installed! many thanks for your help. I'm living in Argelia (north africa). I have configure my MFCR2 according argentina R2 settigs : the test call run perfectly (only warnings and I think that is just debug). but I have many problems and when I run Asterisk-MFCR2. generally in the begging no errors occur. after

Dear Steve and *.* e1r2 developers and users, now MFCR2 is successfully installed! many thanks for your help. I'm living in Argelia. I have configure my MFCR2 according argentina R2 settigs. (look at the end of the message) the testcall run perfectly (only warnings and I think that is just debug). but I have many problems and when I run Asterisk-MFCR2, generally in the begging no errors

Hello I'm using mfcr2 support (unicall) in asterisk 1.4. Everything is working fine, asterisk can answer calls. But after some random period of time mfcr2 module stuck. When I make a call to my * box I can hear only signal of getting caller ID ("tritirirti" - like jumping on :) ) and connection is terminated by my telecom operator. When everything I ok after few seconds of this

Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a astunicall-1.4 setup with a te110p to a nortel pbx in Mexico. (Hate R2!). This is what I get when trying to call to * box using testcall: ./testcall Chan 31, class 'mfcr2', variant 'mx,20,4', end 0, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 31: Call control(9) MFC/R2

Does anybody have a working Asterisk server with Unicall using MFCR2 in Brazil? Were having problems. It seems SPANDSP never detect the tones from the telco. Im using brazil protocol variant. Im having lots of problems to find out why spandsp seems to not detect the MF tones. We send the first digit, the telco says they receive it, and respond with the proper signal to ask for the next digit, we

Hi List, I'm trying to connect Asterisk with a Nortel Meridian using an E1 with MRFR2 signaling. I've connected both cards, and compiled all the required software... The problem is every call (outgoing or incoming) got dropped, complaining about some "T1 timed out" Only for testing purposes I'm using an application called testcall included on the lib-unicall package, and

Is there any way to compile Unicall's libraries (mfcr2, spandsp, chan_unicall, etc) for Asterisk 1.4? BarZ

Hi Bruno, actually vicidial is working on top of asterisk, vicidial doesn't know what asterisk using in layer 2. SS7, ISDN stack, Unicall/mfcr2 is working with asterisk. vicidial uses asterisk application to deliver call center functionalities. Regards, Vidura. ================ Dear Bruno, I had the experience of using the Vcidial with the boards of Digivoice. It worked very well!

Hi, would be glad to be contributing with this question to all the comunity. I?m having a weird issue, suddenly I get channels on IDLE 0x00 state until I do a dahdi restart. Trying to do a call trace to see whats going on deeper, I get surprised when tried to open the .call file to see the log this is incomplete. This is what I see [root at localhost telefonica]# nano