[asterisk-users] Asterisk - Vtiger integration
Dear, I have Asterisk 1.8 (installed with Elastix 2.4) and I want to integrate a Vtiger 6.5 server. In my PBX I have Asterisk 1.8, Java 1.4 and I have not Java Jetty. What are the requirements in the Asterisk server in order to install the VtigerAsteriskConnector package and then integrate the services. Thanks a lot. -- _ -- 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] Custom extension: dial a queue
No, Local/queue/ don't work at all :( 2012/2/6, Danny Nicholas da...@debsinc.com: Local/queue/? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Monday, February 06, 2012 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Custom extension: dial a queue Dear, I need to create a custom device extension in order to dial a local queue. Suppose my queue number is , how can fill the Dial field from the custom extension ??? Because if I put just or Local/, I don't succeed. Thanks a lot -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] Custom extension: dial a queue
No :( 2012/2/6, Danny Nicholas da...@debsinc.com: Queue()? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Monday, February 06, 2012 1:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Custom extension: dial a queue No, Local/queue/ don't work at all :( 2012/2/6, Danny Nicholas da...@debsinc.com: Local/queue/? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Monday, February 06, 2012 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Custom extension: dial a queue Dear, I need to create a custom device extension in order to dial a local queue. Suppose my queue number is , how can fill the Dial field from the custom extension ??? Because if I put just or Local/, I don't succeed. Thanks a lot -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] Spy just a range of extensions
Dear Bakko, I use this lines in order to listen and whisper:
[custom-spy]
; Listen
exten = _*84.,1,Set(SPY=${EXTEN:3})
exten = _*84.,n,NoOp(spy an agent: ${SPY})
exten = _*84.,n,ChanSpy(Agent/${SPY},q)
exten = _*84.,n,Hangup
; Whisper
exten = _*85.,1,Set(SPY=${EXTEN:3})
exten = _*85.,n,NoOp(whisper an agent: ${SPY})
exten = _*85.,n,ChanSpy(Agent/${SPY},w)
exten = _*85.,n,Hangup
Where do I have to add the e(ext) option here ???
Thanks a lot again !!!
2011/8/22 bakko asannu...@gmail.com
Hi Alejandro,
if you use 1.6.2.X look at e(ext) option
With this option you can spy only the extensions you define, separate with
: delimiter.
Example:
exten - Chanspy,1,(all,e(9000:9001:**9002:9002)
I don't test this option but I think work.
Regards
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[asterisk-users] Spy just a range of extensions
Dear, I have to let some agents from a call center to spy/coach just a range
of extensions. They must not spy extensions from boss and some other
important people from my company.
I have in extensions_additional.conf:
[app-chanspy]
include = app-chanspy-custom
exten = 555,1,Macro(user-callerid,)
exten = 555,n,Answer
exten = 555,n,Wait(1)
exten = 555,n,ChanSpy()
exten = 555,n,Hangup
and in extensions_custom.conf:
[from-internal-custom]
exten = 555,1,Macro(user-callerid)
exten = 555,2,Authenticate(1234)
exten = 555,3,Read(SPYNUM,agent-newlocation)
exten = 555,4,ChanSpy(SIP/${SPYNUM))
How can I let agents to just spy extensions 9000-9500 and no more ???
Special thanks,
Alejandro
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[asterisk-users] Ring delay problem
Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read dialparties.agi use PHP interpreter, so I notice when I use PHP command to execute a .php script, it is too slow !!! So I think the problem es PHP. After that I reinstall php* packages with yum reinstall php*, but I have the same problem: the called extension rings after 20 seconds. Any idea please ??? Really thanks. -- _ -- 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] Ring delay problem
Warrem thanksa lotI'll test next monday and I'll tell you. Regards 2011/8/5 Warren Selby wcse...@selbytech.com On Fri, Aug 5, 2011 at 7:53 AM, Alejandro Cabrera Obed aco1...@gmail.comwrote: Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've had this issue before. Try moving the /etc/php.d/imap.so file out of the /etc/php.d directory and see if that helps. It's been a while but I may have had to restart the machine when I did the file move. It may have also just been a DNS timeout issue, I don't recall the specifics. I believe I used this thread as a reference: http://www.fonality.com/trixbox/forums/trixbox-forums/help/suddenly-everything-slow -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] Disabling echo cancellation by software
Dear, I have Asterisk 1.6 with an E1 Digium card with echo cancellation module. So I need to use just the echo cancellation by hardware and disable the echo cancellation by software. I use DAHDI for my telephony hardware. If the lines involved with echo cancel are: In /etc/dahdi/system.conf: echocanceller=mg2,1-15,17-31 In /etc/asterisk/chan_dahdi.conf: echocancel=yes echocancelwhenbridged=no echotraining=800 What lines do I have to comment or to change the value if I want to disable echo cancellation by software ??? 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
Re: [asterisk-users] Blacklist with *30
Dear Dovis, I'm using Elastix and the dialplan comes with this line: *30,1,Goto(app-blacklist-add,s,1) Any idea ??? Thanks a lot. 2011/5/9 Dovid Bender asteriskus...@dovid.net: Alejandro, What GUI are you using ? I don't think Asterisk comes with *30 to ban calls. Regards, Dovid - Original Message - From: Alejandro Cabrera Obed aco1...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 06, 2011 23:51 Subject: [asterisk-users] Blacklist with *30 Dear, when I dial *30 in order to get instructions to blacklist an extension, Idon't get the menu but I get a new dial tone. What happen please ??? What can I do to solve this ??? 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 -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] Blacklist with *30
Dear, finally I implement the functionality code *94 in order to access the blacklist menu from my own extension and put another extension in the black list of Asterisk. But after blacklisting a given extension, when I call from that extension to my own extension the call always rings, it is not denied by the blacklist. Why could be the problem the blacklist doesn't work ??? Thanks a lot 2011/5/9 Dovid Bender asteriskus...@dovid.net: Try the Elastix forums. - Original Message - From: Alejandro Cabrera Obed aco1...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 09, 2011 15:35 Subject: Re: [asterisk-users] Blacklist with *30 Dear Dovis, I'm using Elastix and the dialplan comes with this line: *30,1,Goto(app-blacklist-add,s,1) Any idea ??? Thanks a lot. 2011/5/9 Dovid Bender asteriskus...@dovid.net: Alejandro, What GUI are you using ? I don't think Asterisk comes with *30 to ban calls. Regards, Dovid - Original Message - From: Alejandro Cabrera Obed aco1...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 06, 2011 23:51 Subject: [asterisk-users] Blacklist with *30 Dear, when I dial *30 in order to get instructions to blacklist an extension, Idon't get the menu but I get a new dial tone. What happen please ??? What can I do to solve this ??? 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
[asterisk-users] Blacklist with *30
Dear, when I dial *30 in order to get instructions to blacklist an extension, Idon't get the menu but I get a new dial tone. What happen please ??? What can I do to solve this ??? 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
[asterisk-users] Reach PSTN from another Asterisk
Dear, we have the following: - Asterisk A with E1 to PSTN connection. - Asterisk B with IAX trunk to Asterisk A - Outgoing routes between Asterisk A and B - Asterisk A with an outgoing route to PSTN with 9|. dial rule How can I reach the PSTN from Asterisk B through Asterisk A ??? 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
[asterisk-users] Asterisk-Asterisk E1 connection
Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both boxes. I need to connect both PBXs with E1/R2 and UTP cable. What are the requirements to deploy the UTP cable ??? Straight-through or crossover ??? What are the pinouts in both peers ??? 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
[asterisk-users] Setting two E1 cards
Dear, I always had one E1 card with one span, so I've never had any problem in set it up through /etc/dahdi/sustem.conf and /etc/asterisk/chan_dahdi.conf because I put span=1. But now I have a PBX with two E1 cards with 4 span (8 span in total). How do I have to define both card in system.conf and chan_dahdi.conf, and how do I have to refer each span to the corresponding card ??? 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
[asterisk-users] E1 channels real time monitoring
Dear, I have an Asterisk PBX with two E1 cards: Digium TE122 and Sangoma A101D. Sangoma card has SNMP support but Digium card has not, and also SNMP does't give me ral time information. Within CLI Asterisk I execute dahdi show channels but I don't get information about channels usage. What is the best way to have real time monitoring of E1 channels usage and status ??? 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
[asterisk-users] Net2Phone SIP trunk problem
Dear, I have this scenario: - PBX Asterisk 1.6.2.10 with private IP 192.168.0.10 - Behind a Cisco ASA firewall that connects to Internet - SIP trunk to Net2Phone with these parameters (nat=no): host=200.58.113.60 username=DOLLY secret=123456 port=5060 type=peer dtmfmode=rfc2833 disallow=all allow=alawulaw nat=no canreinvite=no qualify=yes -Softphones Xlite The PBX can't register to Net2Phone, and no calls are made and this is the log: -- Executing [...@macro-dialout-trunk:20] NoOp(SIP/9004-0008, Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20) in new stack -- Executing [...@macro-dialout-trunk:21] Goto(SIP/9004-0008, s-CHANUNAVAIL,1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-chanunav...@macro-dialout-trunk:1] Set(SIP/9004-0008, RC=20) in new stack -- Executing [s-chanunav...@macro-dialout-trunk:2] Goto(SIP/9004-0008, 20,1) in new stack -- Goto (macro-dialout-trunk,20,1) -- Executing [...@macro-dialout-trunk:1] Goto(SIP/9004-0008, continue,1) in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [conti...@macro-dialout-trunk:1] GotoIf(SIP/9004-0008, 1?noreport) in new stack -- Goto (macro-dialout-trunk,continue,3) -- Executing [conti...@macro-dialout-trunk:3] NoOp(SIP/9004-0008, TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to other trunks) in new stack -- Executing [conti...@macro-dialout-trunk:4] Set(SIP/9004-0008, CALLERID(number)=9004) in new stack What can be the problem please ??? 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
[asterisk-users] Asterisk and RAID
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with four HD's available, using CentOS as the OS. What's the best RAID type recommendation ??? RAID 1 or RAID 5 ??? Regards 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
[asterisk-users] Vicibox vs VicidialNow
Dear all, I need a call center asterisk's based solution and I see there are two important solution for 120+ agents: VicidialNow and ViciBox Can you tell me the difference between these open source call center solution please ??? Special thanks 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
[asterisk-users] Handling DTMF for number 4
Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for mobile phone calls coming from a GSM Gateway. All the components are set up in DTMFMODE = RFC2238, and so when the caller from mobile touches the IP phone LAN extension, the call is succesfully established. Everything is OK except for the DTMF for number 4, because if the caller from mobile dial 1004 or 1014 extensions -which have the number 4- the calls are errouneosly established with extension 1000. I live in Argentina, but I don't know if the DTMF frequencies are the same than other countries or I have to make a change in somewhere. Can be a problem with the detection of DTMF for number 4 in Asterisk ??? 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
Re: [asterisk-users] Handling DTMF for number 4
Thanks Gareth, when you say that I can choose INBAND for DTMF MODE in the GSM Gateway, that implies that the DTMF MODE of the Asterisk extension registered for the GSM Gateway has to be set to INBAND too or can it remain in RFC2238 ??? Because I have all my Asterisk extensions and IP telephones set up with DTMFMODE = RFC2238 by now, and I can't understand if you suggest me I change the DTMFMODE from RFC2238 to INBAND just in the GSM Gateway or everywhere. Thanks again. 2010/6/28 Gareth Blades list-aster...@skycomuk.com: Alejandro Cabrera Obed wrote: Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for mobile phone calls coming from a GSM Gateway. All the components are set up in DTMFMODE = RFC2238, and so when the caller from mobile touches the IP phone LAN extension, the call is succesfully established. Everything is OK except for the DTMF for number 4, because if the caller from mobile dial 1004 or 1014 extensions -which have the number 4- the calls are errouneosly established with extension 1000. I live in Argentina, but I don't know if the DTMF frequencies are the same than other countries or I have to make a change in somewhere. Can be a problem with the detection of DTMF for number 4 in Asterisk ??? Thanks a lot Alejandro The gsm gateway would be performing the DTMF detection and just sending on what it detected as you have the DTMFCODE set as RFC. Maybe if you set the DTMF mode to INBAND it may pass the audio straight through and allow asterisk to detect it. The problem is that mobile calls are heavily compressed so any entered digits are converted to tones once the information reaches the network operator. As the call is gong back over a mobile connection the DTMF is compressed which results in unreliable detection. -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] IVR extension dialing error
Dear, just a short question: If I use G.711a and G.711b codecs between the Portech GSM Gateway and Asterisk 1.4.23, what DTMF mode is better to use in both sides if a mobile phone call the GSM Gateway in order to contact an internal IP extension (Mobile to LAN scenario): RFC2238 Inband SIP INFO What are the requisites to choose among them Thanks in advance Alejandro 2010/6/18 Danny Nicholas da...@debsinc.com: I would definitely change the prompt from 1 to 0. It is not an advisable practice to have an IVR selection that can be misinterpreted like this. Assuming that all of your extensions are in 1000-1999, 2 for the operator would be just as good; the important thing is that you don't have a single digit extension 1. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Friday, June 18, 2010 7:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR extension dialing error Hi, I tell you I've made some calls from a land-phone to my IVR in order to avoid the possible poor quality of cell phone's DTMF, and when I called extension 1003 I was connected to extension 1000 againthe same error. My IVR says dial 1 to connect to operator or dial the extension in case you know.and my extension ranges is 1000-1999, so I think it could be a problem that extensions and IVR option start with the same digit: 1. When I'll be at work I'm thinking in modify the IVR speech in order to say dial 0 to connect to operator., and not dial 1 to connect to operator, so IVR option and extensions will not start with the same digit. Do you think this may be the problem ??? Thanks a lot and sorry for my interruption. Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: According to this link http://www.smallnetbuilder.com/content/view/30469/82/1/2/ You probably want to make 80 be 120. This is a millisecond delay value, so the 500 value is a give it up proposition; 200 might be doable for your outliers. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Thursday, June 17, 2010 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR extension dialing error OK, now I understand..but just one more question...In the DTMF settings tab from the GSM gateway manager I have this line: Mobile DTMF debounce: 80 (range: 40 ~ 500, default: 80 ) step: 10ms What does this setting really mean and do I have to modify the current value ??? Final thanks :) 2010/6/17 Zeeshan Zakaria zisha...@gmail.com: I once setup a callback system for someone and we had these DTMF issues on constant basis, and all the complains were from cell phone users. At that time I found out that even my own cellphone would not DTMF correctly from certain locations, including my home, but would work perfectly fine from my work location. Probably times of the day matters too, but yes, calling from cell phones does result in DTMF issues, and the reason is that it is just the audio signals, which get distorted based on various factors like the signal strength, cell tower transmission quality, transcodings, etc. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-17 11:25 AM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Danny, so you say it's a problem of the cell phone and not the Astreisk or GSM Gateway ??? OK, in this case if I call from a fixed phone (not a cell phone) to the IVR, the DTMF quality problem will not be presentthis may be a good test, isn't it ??? Or do you suggest another test I can implement ??? Thanks again Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: The physical location of the phone (access to towers) can vastly affect the quality of DTMF pass... -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocati... -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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
Re: [asterisk-users] Asterisk distribution for a Call Center
Thank you for your comments and suggestions !!! Now I will start to read about the different products you mentioned, and take a decission. By the way, a friend of mine suggest to me Trixbox Pro Call Center Edition (paying some $$$) because he says this product has several features directed for a call center like ACD, real-time status of E1 and PRI lines and a so more I don't know they are in Asterisk Now or Vicidialnow. Regards, Alejandro 2010/6/23 Carlo Taguinod cvtagui...@gmail.com: VicidialNOW (http://vicidialnow.org/) On Wed, Jun 23, 2010 at 2:21 AM, Alejandro Cabrera Obed aco1...@gmail.com wrote: 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] Asterisk + E1 card
Dear Doug and people, In order to install an E1 Digium card in Asterisk, I've read it's necessary to have DAHDI. But in my Asterisk installation I have ZAPTEL. Is it the same to have zaptel or dahdi in order to put to work my E1 Digium on my server in a plug and play way ??? because I don't want to install dahdi with wget, I prefer to maintain the original zaptel installation. Special thanks Alejandro 2010/6/16 Doug Lytle supp...@drdos.info: Alejandro Cabrera Obed wrote: Is it necessary to install or update any Asterisk/Zaptel/Any extra module or the default installation is good enough to just plug and run the E1 card You'll need to make sure that libpri and dahdi are installed and configured. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] 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
Re: [asterisk-users] IVR extension dialing error
Hi, I tell you I've made some calls from a land-phone to my IVR in order to avoid the possible poor quality of cell phone's DTMF, and when I called extension 1003 I was connected to extension 1000 againthe same error. My IVR says dial 1 to connect to operator or dial the extension in case you know.and my extension ranges is 1000-1999, so I think it could be a problem that extensions and IVR option start with the same digit: 1. When I'll be at work I'm thinking in modify the IVR speech in order to say dial 0 to connect to operator., and not dial 1 to connect to operator, so IVR option and extensions will not start with the same digit. Do you think this may be the problem ??? Thanks a lot and sorry for my interruption. Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: According to this link http://www.smallnetbuilder.com/content/view/30469/82/1/2/ You probably want to make 80 be 120. This is a millisecond delay value, so the 500 value is a give it up proposition; 200 might be doable for your outliers. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Thursday, June 17, 2010 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR extension dialing error OK, now I understand..but just one more question...In the DTMF settings tab from the GSM gateway manager I have this line: Mobile DTMF debounce: 80 (range: 40 ~ 500, default: 80 ) step: 10ms What does this setting really mean and do I have to modify the current value ??? Final thanks :) 2010/6/17 Zeeshan Zakaria zisha...@gmail.com: I once setup a callback system for someone and we had these DTMF issues on constant basis, and all the complains were from cell phone users. At that time I found out that even my own cellphone would not DTMF correctly from certain locations, including my home, but would work perfectly fine from my work location. Probably times of the day matters too, but yes, calling from cell phones does result in DTMF issues, and the reason is that it is just the audio signals, which get distorted based on various factors like the signal strength, cell tower transmission quality, transcodings, etc. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-17 11:25 AM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Danny, so you say it's a problem of the cell phone and not the Astreisk or GSM Gateway ??? OK, in this case if I call from a fixed phone (not a cell phone) to the IVR, the DTMF quality problem will not be presentthis may be a good test, isn't it ??? Or do you suggest another test I can implement ??? Thanks again Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: The physical location of the phone (access to towers) can vastly affect the quality of DTMF pass... -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocati... -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] IVR extension dialing error
Dear all, I have a GSM Gateway that let me connect from cell phones to the Asterisk's IVR I've created. The IVR let dial any extension you know, so you can dial to the range 1000-1050. When the cell phones are in the metropolitan area evertythin is correct, you dial 1000 and you call to 1000 extension. But when the cell phones are in other states from my country, sometimes you dial 1000 and call to 1005 or 1023 or any other extension than you want to call. Please can you help me in this problem ??? Any idea to avoid this behaviour ??? 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
Re: [asterisk-users] IVR extension dialing error
Danny, so you say it's a problem of the cell phone and not the Astreisk or GSM Gateway ??? OK, in this case if I call from a fixed phone (not a cell phone) to the IVR, the DTMF quality problem will not be presentthis may be a good test, isn't it ??? Or do you suggest another test I can implement ??? Thanks again Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: The physical location of the phone (access to towers) can vastly affect the quality of DTMF passed to *. Look at the CDR for a phone on good and bad calls. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Thursday, June 17, 2010 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] IVR extension dialing error Dear all, I have a GSM Gateway that let me connect from cell phones to the Asterisk's IVR I've created. The IVR let dial any extension you know, so you can dial to the range 1000-1050. When the cell phones are in the metropolitan area evertythin is correct, you dial 1000 and you call to 1000 extension. But when the cell phones are in other states from my country, sometimes you dial 1000 and call to 1005 or 1023 or any other extension than you want to call. Please can you help me in this problem ??? Any idea to avoid this behaviour ??? 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 -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] IVR extension dialing error
OK, now I understand..but just one more question...In the DTMF settings tab from the GSM gateway manager I have this line: Mobile DTMF debounce: 80 (range: 40 ~ 500, default: 80 ) step: 10ms What does this setting really mean and do I have to modify the current value ??? Final thanks :) 2010/6/17 Zeeshan Zakaria zisha...@gmail.com: I once setup a callback system for someone and we had these DTMF issues on constant basis, and all the complains were from cell phone users. At that time I found out that even my own cellphone would not DTMF correctly from certain locations, including my home, but would work perfectly fine from my work location. Probably times of the day matters too, but yes, calling from cell phones does result in DTMF issues, and the reason is that it is just the audio signals, which get distorted based on various factors like the signal strength, cell tower transmission quality, transcodings, etc. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-17 11:25 AM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Danny, so you say it's a problem of the cell phone and not the Astreisk or GSM Gateway ??? OK, in this case if I call from a fixed phone (not a cell phone) to the IVR, the DTMF quality problem will not be presentthis may be a good test, isn't it ??? Or do you suggest another test I can implement ??? Thanks again Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: The physical location of the phone (access to towers) can vastly affect the quality of DTMF pass... -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocati... -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] Asterisk + E1 card
Dear all, I have to install an E1 card in my Asterisk 1.4.23 server and here is my short question: Is it necessary to install or update any Asterisk/Zaptel/Any extra module or the default installation is good enough to just plug and run the E1 card 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
[asterisk-users] Codec G.129 A vs A/B
Dear all, I've read that Asterisk supports only the G.729 A audio codec. I have several Grandstream IP phones with G.729 A/B codec implementation. Does G.729 A/B mean both version A and version B, or A/B is a new version different from A and B and it's not supported by Asterisk ??? 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
[asterisk-users] Inbound route question
Dear, I have an Asterisk PBX with 3 SIP extensions (1000, 1001 and 1002) and a GSM Gateway with SIP extension . Two cell phones call to the GSM Gateway number and after that they get a ring tone to dial to the SIP extensions. Is it possible to consider the GSM Gateway SIP extension as an incoming call to the Asterisk PBX and so create an inbound route that point: GSM Gateway DID: - IVR in order to point all incoming cell phone calls to my existing IVR ??? 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
Re: [asterisk-users] Inbound route question
But suppose the cell phones DID number is: 11654321 and the GSM Gateway extension has DID number: Which is the DID number I have to use in the inbound route I create to point to the IVR ??? Thanks again. 2010/4/26 Danny Nicholas da...@debsinc.com: I must be missing something because this sounds REAL simple - just dial 1000, 1001 or 1002 from dialplan or do a Goto to the IVR context. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Monday, April 26, 2010 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inbound route question Dear, I have an Asterisk PBX with 3 SIP extensions (1000, 1001 and 1002) and a GSM Gateway with SIP extension . Two cell phones call to the GSM Gateway number and after that they get a ring tone to dial to the SIP extensions. Is it possible to consider the GSM Gateway SIP extension as an incoming call to the Asterisk PBX and so create an inbound route that point: GSM Gateway DID: - IVR in order to point all incoming cell phone calls to my existing IVR ??? 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 -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] G.711a or G.711u ???
Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! 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
[asterisk-users] Context vs. Custom Context
Dear all, if I use the CustomContext module in Asterisk in order to create new customized contexts for my extensions to managed outbound/inbound calls, do these custom contexts replace the original context defined in sip.conf, like context=from-internal ??? In other words, does a custom context have a bigger priority than context ??? 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
Re: [asterisk-users] Context vs. Custom Context
Yes, Custom Context is a module from FreePBX in order to define calling routes. Thanks. 2010/3/22 Leif Madsen leif.mad...@asteriskdocs.org Alejandro Cabrera Obed wrote: Dear all, if I use the CustomContext module in Asterisk in order to create new customized contexts for my extensions to managed outbound/inbound calls, do these custom contexts replace the original context defined in sip.conf, like context=from-internal ??? In other words, does a custom context have a bigger priority than context ??? That sounds like a module for FreePBX or some other GUI. A context in Asterisk is just a context. There are no weights. If you define the same context twice you will likely get some sort of WARNING on the Asterisk console I think. Leif. -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] Adding an external dial code
Dear all, I have Asterisk managed by a FreePBX web console, and I want to add an external dial code, in order to dial 9 to get external line/tone for outgoing calls to the GSM network through my GSM gateway. Where from Asterisk/FreePBX can I setup this feature ??? 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
[asterisk-users] Outbound route prefixes
Dear all, I use Trixbox as my PBX. Until a couple of days I've installed a GSM Gateway to communicate with our three cellular phones: 15 6422 15 6422 15 6422 The GSM Gateway has just one SIM. I use the Free PBX web interface in order to set up the route and trunk parameters: Trunk: *** Name: SIM1 Peer details: host=10.10.1.2 (IP from GSM Gateway) port=5060 type=peer Outbound route: ** Name: SIM1 Dial patterns: 15 (remember I just want to call our three cellular numbers) My GSM Gateway SIP number is 999. After that, I call 999 from my SIP phone, I get new tone, dial ANY phone number and the call is established. How canI restrict my calls through the GSM Gateway to just our three cellular numbres cited above ??? 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
[asterisk-users] Outbound route prefixes
Dear all, I use Trixbox as my PBX. Until a couple of days I've installed a GSM Gateway to communicate with our three cellular phones: 15 6422 15 6422 15 6422 The GSM Gateway has just one SIM. I use the Free PBX web interface in order to set up the route and trunk parameters: Trunk: *** Name: SIM1 Peer details: host=10.10.1.2 (IP from GSM Gateway) port=5060 type=peer Outbound route: ** Name: SIM1 Dial patterns: 15 (remember I just want to call our three cellular numbers) My GSM Gateway SIP number is 999. After that, I call 999 from my SIP phone, I get new tone, dial ANY phone number and the call is established. How canI restrict my calls through the GSM Gateway to just our three cellular numbres cited above ??? Thanks a lot, Alejandro -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] GSM 6.10 codec for Asterisk
Dear all, I'm planning to buy some IP phones with GSM audio codec support in order to use with an Asterisk SIP server I have implemented and nowsuccessfully running with softphones like Eyebeam and Twinkle. A vendor offer to me the SNOM 300 IP phone, that support GSM 6.10 audio codec. I've looking for GSM 6.10 codec in the web but there is no helpful information. Just I enter the Asterisk CLI console and after running the show codecs command I get the GSM codec as valid. Can you tell me if Asterisk support the GSM 6.10 audio codec ??? What the difference between GSM and GSM 6.10 ??? Special thanks, Alejandro ___ -- 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] Complete neutral Spanish sounds
Dear all, does anybody know about a complete set of neutral Spanish sounds to use in my Asterisk voicemail ??? Because when I get a Spanish sounds package, it always is incomplete. I live in Argentina, so I prefer neutral voices. Special thanks Alejandro ___ -- 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] Asterisk and G.729 codec: short questions
Dear all, I have Trixbox 2.6 (Asterisk 1.4) installed in my voip server. I have the following short questions about the usage of G.729 codec: 1) Does Asterisk have installed the G.729 codec by default ??? 2) If I don't want to pay for a codec license, using Asterisk in pass-through mode for G.729 voice communications, do I just have to download the open source version of the G.729 codec or can I use the one coming in Asterisk ??? 3) If I use G.729 for voice communications and GSM for voice mail sounds, does Asterisk execute trascoding ??? Really thanks Alejandro ___ -- 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] Sounds format: GSM to G.729
Dear all, I have an Asterisk SIP PBX using GSM codec at peers and in voicemail sounds files (I have Spanish sounds). But now I have a problem because I have to use G.729 mandatory at peers, and I have GSM in voicemail sound files. I can't let Asterisk do trascoding because I have no a DSP in the CPU, and I don't want to degrade the PBX performance with trascoding tasks. So how can I trascode sounds file from GSM to G.729 ??? Any Linux package suggestion to do this task ??? Because sounds files in /var/lib/asterisk/sounds are a lot as I see. Thanks a lot Alejandro ___ -- 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] G.729 licence in devices connected to Asterisk
Just a short question: I will have Asterisk using G.729 codec and connected to some voip devices such IP phones (GarndStream) and a GSM gateway (Portech). Do IP phones and GSM gateway include valid G.729 licenses or do I have to pay for them ??? Thanks a lot Alejandro ___ -- 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] Sounds format: GSM to G.729
On Fri, Jun 26, 2009 at 4:21 PM, Kevin P. Fleming kpflem...@digium.comwrote: Alejandro Cabrera Obed wrote: Because sounds files in /var/lib/asterisk/sounds are a lot as I see. If you are using the Spanish sounds distributed by Digium, they are already available in G.729 format from downloads.asterisk.org. Thanks Kevin, so If I use G.729 in sound files, IP phones and Asterisk and I not need any trascoding to the PSTN, can I use the codec for free absolutely ??? Thanks again. Alejandro ___ -- 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] Sisky to connect Skype to Asterisk
Dear all, I've read some news about Sisky (http://www.yeastar.com/Products/SiSkyEE.asp), a service to interconnect Skype clients with SIP clients. Does anybody test Sisky and can tell me about his experience ??? (Sisky runs on Windows because Skype and its API are more stable on this OS). Regards, Alejandro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with encryption
Dear all, I want to know if anybody has implented an Asterisk server (1.4 or 1.6) with SRTP and SIP with TLS, in order to encrypt both signaling and voice packets. Is it possible ?? And in the affirmative case, does encryption increase the delay and so the voice quality becomes wrong ??? Thanks a lot. Alejandro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with SRTP and SIP with TLS
Dear all, I want to know if anybody has implented an Asterisk server (1.4 or 1.6) with SRTP and SIP with TLS, in order to encrypt both signaling and voice packets. Is it possible ?? And in the affirmative case, does encryption increase the delay and so the voice quality becomes wrong ??? Thanks a lot. Alejandro ___ -- 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] GSM codec is a good choice ???
But in my case, I don't need trascoding because every chanel is in GSM and voicemail has gsm sound files. And for the moment, my Asterisk is not connected to the PSTN, so there is no trascoding gsm-to-PCM or to analog. So I think gsm is a good choice for my scenario, do you ??? Thanks a lot !!! On Wed, Feb 25, 2009 at 5:33 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Feb 24, 2009 at 11:16:51PM -0200, David fire wrote: out there is a free for educational and no commercial G729 lib for asterisk you can use it to test in a non-comercial system. For personal use? Maybe. For educational use: not really. The licensing of the Intel codec code are not that nice. And naturally, if you wan ta good speech codec with a high quality and yet good compression, and no extra bagage of patents, your first choice should be Speex. -- 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 ___ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar ___ -- 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] GSM codec is a good choice ???
Thanks for your comment about codecsI tell you I can't use G.711 because I use a WAN link, and this is a wide band codec. Is GSM codec totally free (avoid to pay for any license) ??? Thnks again. On Tue, Feb 24, 2009 at 11:50 AM, Tiago Durante tiagodura...@gmail.com wrote: I'd use alaw/ulaw for everything that's local, gsm or g729 only for remote extensions. On 2/24/09, Philipp Kempgen philipp.kemp...@amooma.de wrote: Alejandro Cabrera Obed schrieb: Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. http://www.utopixnetworks.com.ar/ip_phones_hiperphone_202.php According to the web site the Utopix HiperPhone 202 and 112 support G.711a/u (alaw/ulaw) as well. So why not use G.711a for everything? Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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 -- Sent from my mobile device Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich ___ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar ___ -- 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] GSM codec is a good choice ???
Yes, there is a WAN among the hardphones and my Asterisk server. I know the GSM bitrate is about 31 Kbps. Thanks On Tue, Feb 24, 2009 at 1:41 PM, Olivier oza-4...@myamail.com wrote: 2009/2/24 Alejandro Cabrera Obed aco1...@gmail.com Thanks for your comment about codecsI tell you I can't use G.711 because I use a WAN link, and this is a wide band codec. Is GSM codec totally free (avoid to pay for any license) ??? yes ! Is there a WAN between your hardphones and Asterisk ? Thnks again. On Tue, Feb 24, 2009 at 11:50 AM, Tiago Durante tiagodura...@gmail.com wrote: I'd use alaw/ulaw for everything that's local, gsm or g729 only for remote extensions. On 2/24/09, Philipp Kempgen philipp.kemp...@amooma.de wrote: Alejandro Cabrera Obed schrieb: Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. http://www.utopixnetworks.com.ar/ip_phones_hiperphone_202.php According to the web site the Utopix HiperPhone 202 and 112 support G.711a/u (alaw/ulaw) as well. So why not use G.711a for everything? Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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 -- Sent from my mobile device Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich ___ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar ___ -- 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] GSM codec is a good choice ???
Do you think GSM codec has poor audio quality ??? Because I've made some tests among softphones connected from different cities of my country and the audio was good to me. Maybe GSM is a good choice. On Tue, Feb 24, 2009 at 11:16 PM, David fire ddf...@gmail.com wrote: out there is a free for educational and no commercial G729 lib for asterisk you can use it to test in a non-comercial system. the digium lib is much better. if you have more than 30~60 phones transcoding inst a very good idea. i made my self a test on a core 2 duo 64 bits 2GB of ram a test transcoding more than 90 calls the sound quality was BAD not poor BAD. the digium transcoder is GREATE 0 cpu was gone for transcoding. keep this in mind. David 2009/2/24 Kristian Kielhofner kristian.kielhof...@gmail.com On Mon, Feb 23, 2009 at 6:59 PM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. If I choose G.729A the pass-throu calls among users are OK, but Asterisk can't transcode GSM to G.729A in voicemail calls. My company doesn'y want to pay for a G.729 license, so I'm thinking to buy new IP phones with GSM support, so I have no problem with the voicemail system. Are the IP phone with GSM support a good choice for me ??? (Maybe in the future I need to connect the Asterisk with the PSTN, GSM doesn't matter at this point ???) Really thanks, Alejandro Install the G.729 sound files and make app_voicemail record messages (format=g729) in G729. As long as you don't need meetme or a few other apps that essentially require G.729 transcoding you don't need a license. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar ___ -- 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] GSM codec is a good choice ???
Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. If I choose G.729A the pass-throu calls among users are OK, but Asterisk can't transcode GSM to G.729A in voicemail calls. My company doesn'y want to pay for a G.729 license, so I'm thinking to buy new IP phones with GSM support, so I have no problem with the voicemail system. Are the IP phone with GSM support a good choice for me ??? (Maybe in the future I need to connect the Asterisk with the PSTN, GSM doesn't matter at this point ???) Really thanks, Alejandro ___ -- 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] Short question: CPU hardware requirements for Asterisk
Dear all, just a short question: What is the best CPU hardware requirements (CPU, memory, hard drive) to install Asterisk with SIP/RTP protocol for 100-150 users, and routing the RTP traffic by itself (no direct RTP traffic client-to-client) Special thanks Alejandro ___ -- 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
[asterisk-users] Encrypted IP phone compatible with Asterisk
Dear, I'm looking for IP phones (directly connected to the RJ-45 port from my LAN) that support any level of encryption for use with an Asterisk 1.4 SIP server we have. What branch and type can I use What is the encryption mechanism I can have with this equipments ??? Greetings Alejandro ___ -- 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
[asterisk-users] Asterisk connected to the PSTN vs. a commercial solution
Dear all, now I'm using an Asterisk 1.4.13 SIP server with 50 SIP users and it works very well only in an intranet environment (no connections to the PSTN world). But in the near future, we have to plan a telephone system that works in the intranet (voip) and also it must be connected to the PSTN public network with a T1/E1 trunk, with 200 SIP users aproximately. So at first I have to ways to do that: 1- Continue using Asterisk and adding a T1/E1 interface in order to connect to the PSTN 2- Discard Asterisk and buy a commercial solution, because we have the money My questions are: does Asterisk work in the scenario I've described What is the best solution you can recommend to me ??? Thanks in advance, Alejandro ___ -- 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
[asterisk-users] ZRTP in Asterisk
Dear people, does anybody try the ZRTP patch for Asterisk in order to have ZRTP encrytion among SIP/RTP calls ??? In other words, did anybody succesfully implement ZRTP in Asterisk ??? Any documentation about it ??? Special thanks Alejandro ___ -- 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
Re: [asterisk-users] Asterisk's ZRTP patch
Jeff Peeler wrote: On Fri, 2008-06-27 at 11:36 -0300, Alejandro Cabrera Obed wrote: Dear all, I have Asterisk 1.4.13 as our SIP server and several SIP clients using ZRTP support (with Zfone module in Windows and libzrtp in Linux). People say that it's necessary to use an Asterisk patch in order tu support ZRTP encryption. Is it true ??? Or maybe if I use the last version of Asterisk I have the ZRTP feature included ??? Thnking in advance. Alejandro Yes, it's true. Check out http://zfoneproject.com/prod_asterisk.html. I'm personally interested in this project, but the patches have not been disclaimed so I can't even look at it. Jeff Thanks Jeff...but why I can succesfully use ZRTP with Twinkle-to-Twinkle calls without using any patch ??? This is a question I ask myself. Thanks a lot and I'll be waiting for some news about this topic. Regards Alejandro ___ -- 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
[asterisk-users] Maximum number of SIP peers in Asterisk 1.4
Dear all, I have Asterisk 1.4.13 as a SIP server for my company with 100 peers (I mean users) and everything work fine. I have the following question: what is the maximum number of peers that I can reach with Asterisk ??? I know Asterisk is not a SIP server basically like OpenSER, so I'm confused. Thanks a lot, Alejandro ___ -- 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
[asterisk-users] Asterisk's ZRTP patch
Dear all, I have Asterisk 1.4.13 as our SIP server and several SIP clients using ZRTP support (with Zfone module in Windows and libzrtp in Linux). People say that it's necessary to use an Asterisk patch in order tu support ZRTP encryption. Is it true ??? Or maybe if I use the last version of Asterisk I have the ZRTP feature included ??? Thnking in advance. Alejandro ___ -- 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
[asterisk-users] module reload question
Dear all, I have installed asterisk 1.4.13 and configured all the /etc/asterisk files very well. Always I enter the CLI (with asterisk -r) and when I make a change after that I execute module reload and everything is OK. But a few days ago, without make any change, I execute module reload from within CLI and the terminal turn into black color and the color of the letters was white (exactly the opposite to the normal colors). I think because I get some warning and notice message: [May 12 10:19:10] NOTICE[6265]: cdr.c:1362 do_reload: CDR simple logging enabled. [May 12 10:19:10] NOTICE[6265]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'us' [May 12 10:19:10] WARNING[6265]: res_smdi.c:746 reload: No SMDI interfaces were specified to listen on, not starting SDMI listener. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4090 pbx_load_module: Starting AEL load process. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4097 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 112, Cols: 34-34: Warning! The empty context ael-dundi-e164-canonical will be IGNORED! -- Reloading module 'codec_gsm.so' (GSM Coder/Decoder) [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 120, Cols: 34-34: Warning! The empty context ael-dundi-e164-customers will be IGNORED! [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 128, Cols: 33-33: Warning! The empty context ael-dundi-e164-via-pstn will be IGNORED! [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4105 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-canonical' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-customers' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-via-pstn' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 276-283: The included context 'ael-parkedcalls' cannot be found. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4108 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4110 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4113 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-customers' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-via-pstn' [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4116 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. After that I test the system and it work OK. What can be the problem ??? Is it a normal situation ??? Thanks a lot. Alejandro ___ -- 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] module reload CLI Asterisk question
Dear all, I have installed asterisk 1.4.13 and configured all the /etc/asterisk files very well. Always I enter the CLI (with asterisk -r) and when I make a change after that I execute module reload and everything is OK. But a few days ago, without make any change, I execute module reload from within CLI and the terminal turn into black color and the color of the letters was white (exactly the opposite to the normal colors). I think because I get some warning and notice message like these: [[May 12 10:19:10] NOTICE[6265]: cdr.c:1362 do_reload: CDR simple logging enabled. [May 12 10:19:10] NOTICE[6265]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'us' [May 12 10:19:10] WARNING[6265]: res_smdi.c:746 reload: No SMDI interfaces were specified to listen on, not starting SDMI listener. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4090 pbx_load_module: Starting AEL load process. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4097 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 112, Cols: 34-34: Warning! The empty context ael-dundi-e164-canonical will be IGNORED! -- Reloading module 'codec_gsm.so' (GSM Coder/Decoder) [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 120, Cols: 34-34: Warning! The empty context ael-dundi-e164-customers will be IGNORED! [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 128, Cols: 33-33: Warning! The empty context ael-dundi-e164-via-pstn will be IGNORED! [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4105 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-canonical' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-customers' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-via-pstn' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 276-283: The included context 'ael-parkedcalls' cannot be found. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4108 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4110 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4113 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-customers' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-via-pstn' [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4116 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. After that I test the system and it work OK. What can be the problem ??? Is it a normal situation ??? Thanks a lot. Alejandro ___ -- 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] Customize Music On Hold
Dear all, I have Asterisk 1.4.13 with the default configuration for Music On Hold. I have this in /etc/asterisk/musiconhold.conf: [default] mode=files directory=/var/lib/asterisk/moh and in /var/lib/asterisk/moh I have the default wav files: fpm-calm-river.wav fpm-sunshine.wav fpm-world-mix.wav This way the music on hold works very good. After that I use audacity to export my own MP3 files to WAV, and finally I put them into /var/lib/asterisk/moh and delete the default fpm* wav files. But when we call any other and turn on the HOLD function, the music doesn't work. How can I customize the music on hold files ??? Special thanks. Alejandro ___ -- 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] E-mail date is wrong
Dear all, I'm using Asterisk 1.4.13 with voicemail feature. When anybody receive a voice message, he/she receives a mail with the audio attachment. After that I dial the voicemail number and I hear the envelope message that is correct (America/Argentina/Buenos_Aires) which is GMT-3, but when I see the message date header it is wrong because the date correspond to GMT and not GMT-3. Where can I set the date/time in order to put Asterisk to send message with the correct time ??? (The Linux server date is correctly set). Thanks a lot Alejandro ___ -- 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] Voicemail: afternoon audio file is missing
Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with envelope=yes and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full: File digits/afternoon does not exist in any format [Apr 10 14:09:08] WARNING[12955]: file.c:866 ast_streamfile: Unable to open digits/afternoon (format 0x2 (gsm)): No such file or directory [Apr 10 14:09:08] WARNING[12955]: say.c:409 wait_file: Unable to play message digits/afternoon I use language=es and I use the AsteriskSounds_ES.tar.gz (unpacked in /var/lib/asterisk/sound/es) package in order to get Spanish audio files. What can I do to correct the afternoon file error ??? Special thanks Alejandro ___ -- 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] Control of RTP open ports
Dear all, I have an Asterisk 1.4.13 with SIP protocol and several voip clients (Twinkle, X-Lite and SJPhone). Every call among voip clients pass through the Asterisk server, so there isn't any voip packet client-to-client. Can Asterisk control the RTP open ports the voip clients use ??? Or the RTP open ports depend on the voip clients ??? Special thanks Alejandro ___ -- 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] Off-Topic: Avaya
Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot ___ --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] iptables requirements for SIP
Dear all, I have to implement a linux/iptables firewall between my SIP clients and the Asterisk 1.4.13 SIP server. There is no NAT in my implementation, so in sip.conf I have canreinvite=no. I have iptables 1.3.6 version. Does iptables need any SIP special module or something like this in order to let SIP+RTP work OK ??? Special thanks Alejandro ___ --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 1.4 with LDAP
Anthony Francis wrote: Pepo wrote: Hi friends. How do I can use Asterisk 1.4 with LDAP? I need it because the system must use just one password for each user for everything. A lot of thanks. What exactly in asterisk would your LDAP be authenticating? Sip registrations? Thats a device, not a user. In my case, I have a LDAP service that manages users and passwords where mail and squid users from my LAN authenticate to. So I'll be very happy if I can put Asterisk 1.4 users (defined in sip.conf) to authenticate to my LDAP service. An example of my user definition in sip.conf is: [alejandro] type=friend username=alejandro secret=xxx host=dynamic nat=no canreinvite=no context=company disallow=all allow=gsm allow=speex allow=g726 What can I do this ??? Special thanks Alejandro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 + Presence
Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, do I have to establish any special setting in Asterisk 1.4 ??? Or the presence status (online, offline, away, etc.) is only up to the SIP clients and not up to the Asterisk ??? Really thanks Alejandro ___ --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] Off-Topic: add GSM codec to X-Lite
Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in allow=gsm line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF manual of X-Lite 3.0 say it has GSM builtin support. Do you know what's the matter with X-Lite and GSM ??? Can I add it ??? Really thanks Alejandro ___ --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] Off-Topic: add GSM codec to X-Lite
SIP wrote: Alejandro Cabrera Obed wrote: Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in allow=gsm line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF manual of X-Lite 3.0 say it has GSM builtin support. Do you know what's the matter with X-Lite and GSM ??? Can I add it ??? Really thanks Alejandro ___ --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 It lists GSM on my audio codec settings. Perhaps there's something wrong with your install? Try disabling the Zero Touch bandwidth detection. It has, in the past, interfered with my selection of codecs. N. Thanks for your support...I've uninstalled my X-Lite 3.0 softphone and after that I've downloaded the X-Lite 3.0 again from the official web site. But when I go to audio codecs settings, the GSM codec is not present. I disable the zero touch bandwith detection and restart the softphone, but the GSM codec is not present at all. Any idea ??? Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4: encryption support
Dear all, I have Asterisk 1.4.13 and I need to use encryption among Asterisk and my SIP users, and with the RTP data interchanged among users. I prefer the use of ZRTP/SRTP because we use Twinkle and X-Lite/Zfone as our voip clients and they support these encryption mechanism. My question is: do I have to enable any encryption support in Asterisk 1.4.13 ??? Or Asterisk has native encryption support ??? Thanks a lot Alejandro ___ --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] Error: 603 declined
I have Asterisk 1.2.13 installed as a Debian package and I edit only sip.conf and extensions.conf in this way: sip.conf: [general] realm=work.com.ar ; Realm for digest authentication bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes [user1] type=friend username=user1 secret=xxx host=dynamic context=work [user2] type=friend username=user2 secret=xxx host=dynamic context=work extensions.conf: [work] exten = ,1,Dial(SIP,user1) exten = 1112,1,Dial(SIP,user2) When we use Twinkle as our SIP client, user1 calls user2 dialing and user2 calls user1 dialing 1112, we get this error: Line 1 Call failed - 603 declined.so I can make a call. In Asterisk I debug the channel and I get this log: voip*CLI debug channel 1 No such channel 1 Debugging on new channels is enabled -- Executing Dial(SIP/user1-08148450, SIP|user2) in new stack Oct 9 12:52:41 WARNING[3478]: app_dial.c:1024 dial_exec_full: Dial argument takes format (technology/[device:]number1) == Spawn extension (sintys, 1112, 1) exited non-zero on 'SIP/user1-08148450' Oct 9 12:52:41 WARNING[3453]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x81508e8', 10 retries! What is the problem ??? Any help please ??? Thanks a lot Alejandro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.13 and presence
Dear people, is it possible to have presence using Asterisk 1.2.13 / SIP with Linux/Debian Etch??? I'd like to see if my intranet contacts are available, busy, disconnected Thanks a lot Alejandro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Errors: Too many SIP headers and Unknown SDP media type in offer: video 10702 RTP/AVP 34 31
Dear all, I have Asterisk 1.2.13 running OK with Twinkle clients, they can talk very well using SIP. I have a Jabber server running OK and the clients use PSI client for chat succesfully. Now I'm using Wengophone 2.1.1 in order to unify voip+IM services. The users can logon OK in SIP and Jabber, they get the online status presence, but they CAN'T talk and chat among them. From the Asterisk console I can see these errors I can't understand: Sep 7 11:28:06 WARNING[2571]: chan_sip.c:3399 parse_request: Too many SIP headers. Ignoring. Sep 7 11:28:06 WARNING[2571]: chan_sip.c:3399 parse_request: Too many SIP headers. Ignoring. Sep 7 11:28:06 WARNING[2571]: chan_sip.c:3399 parse_request: Too many SIP headers. Ignoring. Sep 7 11:28:06 WARNING[2571]: chan_sip.c:3399 parse_request: Too many SIP headers. Ignoring. Sep 7 11:28:06 WARNING[2571]: chan_sip.c:3399 parse_request: Too many SIP headers. Ignoring. Sep 7 11:28:06 WARNING[2571]: chan_sip.c:3399 parse_request: Too many SIP headers. Ignoring. Sep 7 11:28:06 WARNING[2571]: chan_sip.c:3399 parse_request: Too many SIP headers. Ignoring. Sep 7 11:28:06 WARNING[2571]: chan_sip.c:3399 parse_request: Too many SIP headers. Ignoring. Sep 7 11:28:06 WARNING[2571]: chan_sip.c:3399 parse_request: Too many SIP headers. Ignoring. Sep 7 11:28:06 WARNING[2571]: chan_sip.c:3399 parse_request: Too many SIP headers. Ignoring. Sep 7 11:28:06 WARNING[2571]: chan_sip.c:3602 process_sdp: Unknown SDP media type in offer: video 10702 RTP/AVP 34 31 What can be the problem ??? Thanks Alejandro -- Ing. Alejandro Cabrera Obed Interconexion SINTyS Sistema de Identificación Nacional Tributario y Social Consejo Nacional de Coordinación de Políticas Sociales Presidencia de la Nación Julio A. Roca 782 - Piso 5 Ciudad Autónoma de Bs. As. Tel: (54 11) 4343-0181/89 interno 5172 4334-3676 4342-5648 [EMAIL PROTECTED] NOTA DE RESPONSABILIDAD: -- Este mensaje proviene de Internet,tome los recaudos necesarios en su manejo. El contenido del presente mensaje y sus adjuntos es privado, estrictamente confidencial y exclusivo para su destinatario, pudiendo contener información protegida por normas legales y de secreto profesional. Bajo ninguna circunstancia su contenido puede ser transmitido o revelado a terceros ni divulgado en forma alguna. En consecuencia de haberlo recibido solicitamos contactar al remitente y eliminarlo de su sistema. -- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] VoIP+IM with Asterisk+Jabber
People, I have an Asterisk 1.2 server and a Jabber server in different hosts. I need to implement voip+presence+instant messaging knowing that Asterisk does not support presence+IM.So is it possible to use a softphone client (Gaim, X-Lite, etc.) to give to my users voip+presence+IM connecting to the Asterisk and Jabber servers at the same time ??? Thanks a lot Alejandro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with IM (instant messaging)
Hi people, I have Asterisk 1.2.13/DebianEtch as my VoIP server, using SIP. I need to use IM (instant messaging) among X-Lite clients, but when I send a message to any other client I get the error Error: method not allowed. I read Asterisk does not support instant messaging, so.What's the best way to have instant messaging with Asterisk ??? Thanks a lot. Alejandro ___ --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] VoIP encryption with SIP and IAX
Dear all, I have an Asterisk server with SIP and IAX softphones clients, and I need to encrypt the voip calls among them: *For SIP clients I use Twinkle which implements the ZRTP/SRTP encryption mechanism client-2-client; I read it's the better security mechanism nowadays created by Phill Zimmerman who created PGP. *For IAX clients I used Kiax but I don't know exactly if there is any encryption mechanism for this protocol. Two short questions: 1) Do you think ZRTP/SRTP is the best option to encrypt SIP voip calls ??? 2) What is the best way to encrypt IAX voip calls ??? Really thanks Alejandro ___ --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] VoIP + IM unified client
Dear all, I have a Debian/Asterisk server and I connect several softphones using SIP in a first test and IAX in a second test. They work OK in both cases; I use Twinkle client for SIP conversations and Kiax for IAX. But now I want to have IM also, I mean a voip client with a chat messenger incorporated, always using Asterisk. My questions are: 1) Do I have to add some module/package to my Asterisk in order to have IM ??? 2) What SIP+IM client do you recommend to me ??? 3) And what IAX+IM client do you recommend to me ??? Thanks in advance, Alejandro ___ --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] IAX implementation question
People, I've setup Asterisk in a basic mode with SIP protocol. In the future I wanna connect several offices each one with an own Asterisk server, using IAX because I read it has no firewalling problems using just one UDP port for control and data -aming other advantages- . SIP has NAT problems I know. Do you recommend the use of IAX instead of SIP for users and among several Asterisk's ??? Does the IAX implementation take any extra considerations than SIP ??? Any initial guide for IAX - Asterisk configuration ??? Thanks a lot, Alejandro ___ --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] Destar web interface problem
People, I have a Debian box with Asterisk and I've installed the Destar package in order to get web managing of my voip system. After I installed Destar, it runs on localhost:8080, but my server does not have X-Window to access to it so I can engter the web interface.. So how can I change localhost:8080 to IP_ASTERISK:8080 in order to access Destar via web from another PC ??? Really thanks, Alejandro ___ --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] Asterisk without PSTN interface cards
People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with apt-get the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? 2) Do I have to configure a dummy PSTN interface in my case ?? And if you have a debian-asterisk howto, I really thank you. Regards, Alejandro ___ --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
Re: [asterisk-users] Asterisk without PSTN interface cards
Tzafrir Cohen wrote: On Tue, Apr 10, 2007 at 10:58:45AM -0300, Alejandro Cabrera Obed wrote: People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with apt-get the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? http://packages.debian.org/asterisk (Hey, Etch is out! oldstable no longer has Asterisk 0.1 ;-) As you can see, there is 1.2.16 (and soon 1.2.17, I've already asked to upload it) in Sid, and 1.4.2 in Experimental . Etch has 1.2.13 . Alternatively, try: deb http://updates.xorcom.com/rapid etch main which has some backports of Sid packages. 2) Do I have to configure a dummy PSTN interface in my case ?? You need the zaptel module ztdummy. As you just need ztdummy and not a real zaptel, there's really no reason to use latestgreatest bleeding-edge zaptel. If you added my packages source from above: apt-get install zaptel zaptel-modules-`uname -r` /etc/init.d/zaptel start If you have just the standard Etch sources, the procedure is a bit more complicated, because you have to generate the package zaptel-modules for your kernel: apt-get install zaptel zaptel-source build-essential # maybe you need to also explicitly install linux-headers-`uname -r` # to build and install the zaptel-modules package for your kernel: # (Will probably fetch the proper linux-headers package as well) m-a a-i zaptel /etc/init.d/zaptel start In both cases You should get an error from ztcfg because there's no zaptel.conf, but just ignore it, as you don't need ztcfg for ztdummy. To make that error disappear you can run: touch /etc/zaptel.conf And if you have a debian-asterisk howto, I really thank you. As usual with Debian, start from /usr/share/doc/PACKAGE/README.Debian . Two other potentially-useuful packages in our repository: freepbx # though still a bit broken. maybe try # 'freepbx-common freepbx-modules' asterisk-config-simple Maybe they'll also help you getting started. Dear people, thanks for your help...I appreciatte it a lot. But one more question please: I have a Debian host base with vserver support (virtual machines, I use them for running squid, postfix and a lot of services without problems) I?ve just installed Asterisk in a new vserver from Debian Etch repositories and I get this error: Setting up zaptel (1.2.11.dfsg-1) ... mknod: `/dev/zap/ctl': Operation not permitted dpkg: error processing zaptel (--configure): subprocess post-installation script returned error exit status 1 After that I see the content of /dev/zap and there is nothing at all. Any idea ??? Can I continue without this device if I use only softphones ??? Thanks again Alejandro -- Alejandro Cabrera Obed Interconexion SINTyS Sistema de Identificacio'n Nacional Tributario y Social Consejo Nacional de Coordinacio'n de Poli'ticas Sociales Presidencia de la Nacio'n Julio A. Roca 782 - Piso 5 Ciudad Auto'noma de Bs. As. Tel: (54 11) 4343-0181/89 interno 5172 4334-3676 4342-5648 [EMAIL PROTECTED] NOTA DE RESPONSABILIDAD: -- Este mensaje proviene de Internet,tome los recaudos necesarios en su manejo. El contenido del presente mensaje y sus adjuntos es privado, estrictamente confidencial y exclusivo para su destinatario, pudiendo contener informacio'n protegida por normas legales y de secreto profesional. Bajo ninguna circunstancia su contenido puede ser transmitido o revelado a terceros ni divulgado en forma alguna. En consecuencia de haberlo recibido solicitamos contactar al remitente y eliminarlo de su sistema. -- ___ --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] Asterisk: recommended installation
Dear all, I'll implement a VoIP system using Asterisk + SIP with softphones; I need to connect LAN and VPN users (about 100-150). What version/installation of asterisk do you recommend tyo me ??? Does [EMAIL PROTECTED] or Trixbox match to my scenario By the way, I use Debian Etch as OS server. Really thanks. Alejandro -- Alejandro Cabrera Obed Interconexion SINTyS Sistema de Identificación Nacional Tributario y Social Consejo Nacional de Coordinación de Políticas Sociales Presidencia de la Nación Julio A. Roca 782 - Piso 5 Ciudad Autónoma de Bs. As. Tel: (54 11) 4343-0181/89 interno 5172 4334-3676 4342-5648 [EMAIL PROTECTED] NOTA DE RESPONSABILIDAD: -- Este mensaje proviene de Internet,tome los recaudos necesarios en su manejo. El contenido del presente mensaje y sus adjuntos es privado, estrictamente confidencial y exclusivo para su destinatario, pudiendo contener información protegida por normas legales y de secreto profesional. Bajo ninguna circunstancia su contenido puede ser transmitido o revelado a terceros ni divulgado en forma alguna. En consecuencia de haberlo recibido solicitamos contactar al remitente y eliminarlo de su sistema. -- ___ --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
