[asterisk-users] transfer problem
Hello, I've tried to transfer a IAX call to a number configured on a
traditional
PBX, but it doesn't work. I have a traditional PBX connected with a zap channel
to Asterisk in the following way:
IAX/SIP client -- Asterisk (FXO) -- (FXS) traditional PBX --- OFFICE
Phones
Asterisk is connected to the PBX with an internal number configured inside it.
In other words i keep an internal line an i connect it to an fxo port of
asterisk while asterisk is connected to internet and from here comes iax calls
to talk with other numbers in the office connected to the traditional PBX.
Well, calls to a SIP clients defined in asterisk works fine, but calls to Zap
clients doesn't work. In the most basic form I do:
exten = _44XX,1,Answer
exten = _44XX,n,Dial(Zap/g1/${EXTEN:2}|20|tTr)
exten = _44XX,n,Hangup
and the console logs for this are:
Executing Dial(SIP/sipuser-081d13f0, Zap/g1/38|20|tTr) in new stack
-- Called g1/38
-- Zap/1-1 answered SIP/ggonzalez-081d13f0
Here Dial cmd do one ring and nothing more, Zap channel has answered but the
number dialed never RING, what is wrong? what i have to do get this working
fine?. Thanks for any help
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[asterisk-users] Dtmf tones and SIP
Hi list, I tried to use DISA in order to get the line when I call with my mobile phone but the system doesn't recognise my DTMF tones when I call to a SIP trunk. Everything is working Ok if I use a ZAP Trunks. I tried to google to find a solution but I wasn't able to find any. Any idea? I'm using trixbox 2 with 4 SIP trunks and a Zaptel TDM 400 card. Bye ___ --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 1.4 and CDR
Hi guys, I have recently installed a Asterisk Server with CDR Call Detail Records. I have installed it over a Asterisk 1.2 , but now It do not run . I have installed it with the following procedure: # yum install ncurses #yum install openh323-devel # yum install mysql-server # yum install mysql # yum install php-gd # yum install php-mysql # yum install mysqlclient10 # yum install zlib # yum install zlib-devel # yum install ncurses-devel Install perl support perl -MCPAN -e install DBD::mysql I compile /usr/src/asterisk-addons as follows: # ./configure # make clean # make install In the file /etc/asterisk/cdr_mysql.conf [global] hostname=localhost dbname=asteriskcdrdb table=cdr password=strongpass user=asterisk port=3306 userfield=1 In the File asterisk-stat define (WEBROOT, http://192.168.190.10/asterisk-stat/;); define (FSROOT, /var/www/html/asterisk-stat-v2/); define (LIBDIR, FSROOT.lib/); define (HOST, localhost); define (PORT, 3306); define (USER, asterisk); define (PASS, strongpass); define (DBNAME, asteriskcdrdb); define (DB_TYPE, mysql); // mysql or postgres define (DB_TABLENAME, cdr); When I compile asterisk-addons it pass very good, but I do not build the file *cdr_addon_mysql.so* Do you have similar problem ?Thanks for your response. Excuseme for my english, it is not my native language. ___ --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] windows mobile 5 softphone for square screen devices
Hi, I've been looking for a good SIP application for Windows Mobile for ages. I found speaQ, but it has the same problem as any other softphone for Windows Mobile. You see, it uses the speaker to output the conversation instead of the phone speaker, you know the one that is used when you make a normal phone call with your WM Mobile PDA/Smartphone. At first I was asking myself if every SIP client developer out there is down right stupid but in the end I found out this is actually Microsoft blocking access to that phone speaker. The claim that allowing the developers to access it would allow for invasion of privacy (like recording phone calls). So unless someone can work around this, softphones for WM will remain quite useless. Timothy. Anton Krall wrote: Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones? ___ --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 ___ --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 + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --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 1.4 and CDR
I had the same issue. I needed to install #yum install mysql-devel. Once I did this the addons compiled the file fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pablo Almido Sent: Wednesday, January 17, 2007 9:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4 and CDR Hi guys, I have recently installed a Asterisk Server with CDR Call Detail Records. I have installed it over a Asterisk 1.2 , but now It do not run . I have installed it with the following procedure: # yum install ncurses #yum install openh323-devel # yum install mysql-server # yum install mysql # yum install php-gd # yum install php-mysql # yum install mysqlclient10 # yum install zlib # yum install zlib-devel # yum install ncurses-devel Install perl support perl -MCPAN -e install DBD::mysql I compile /usr/src/asterisk-addons as follows: # ./configure # make clean # make install In the file /etc/asterisk/cdr_mysql.conf [global] hostname=localhost dbname=asteriskcdrdb table=cdr password=strongpass user=asterisk port=3306 userfield=1 In the File asterisk-stat define (WEBROOT, http://192.168.190.10/asterisk-stat/;); define (FSROOT, /var/www/html/asterisk-stat-v2/); define (LIBDIR, FSROOT.lib/); define (HOST, localhost); define (PORT, 3306); define (USER, asterisk); define (PASS, strongpass); define (DBNAME, asteriskcdrdb); define (DB_TYPE, mysql); // mysql or postgres define (DB_TABLENAME, cdr); When I compile asterisk-addons it pass very good, but I do not build the file cdr_addon_mysql.so Do you have similar problem ?Thanks for your response. Excuseme for my english, it is not my native language. ___ --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] transfer problem
I don't think that the first priority (exten = _44XX,1,Answer) is ok,
have you tried without it?
Try not answering and post what happens.
On 1/17/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello, I've tried to transfer a IAX call to a number configured on a
traditional
PBX, but it doesn't work. I have a traditional PBX connected with a zap channel
to Asterisk in the following way:
IAX/SIP client -- Asterisk (FXO) -- (FXS) traditional PBX --- OFFICE
Phones
Asterisk is connected to the PBX with an internal number configured inside it.
In other words i keep an internal line an i connect it to an fxo port of
asterisk while asterisk is connected to internet and from here comes iax calls
to talk with other numbers in the office connected to the traditional PBX.
Well, calls to a SIP clients defined in asterisk works fine, but calls to Zap
clients doesn't work. In the most basic form I do:
exten = _44XX,1,Answer
exten = _44XX,n,Dial(Zap/g1/${EXTEN:2}|20|tTr)
exten = _44XX,n,Hangup
and the console logs for this are:
Executing Dial(SIP/sipuser-081d13f0, Zap/g1/38|20|tTr) in new stack
-- Called g1/38
-- Zap/1-1 answered SIP/ggonzalez-081d13f0
Here Dial cmd do one ring and nothing more, Zap channel has answered but the
number dialed never RING, what is wrong? what i have to do get this working
fine?. Thanks for any help
___
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--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088
Share your knowledge, use free software.
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[asterisk-users] Erratic Snom MWI lights
Long story short... Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Useragent: snom360/6.5.2 Function: F_RETRIEVE [EMAIL PROTECTED] ~]# asterisk -rx show version Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on 2006-11-17 16:35:22 UTC [gateway] exten = 201,hint,SIP/201 exten = 201,1,Dial(SIP/201|20|tr) exten = 201,2,Voicemail([EMAIL PROTECTED]) exten = 201,3,Hangup -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --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] FW: Realtime Voicemail Password Change Not Working
I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, enter new password ok, re-enter new password ok, password has been changed There are no entries in the mysql.log setting the new password in the database. How can I isolate between asterisk, realtime driver, and mysql? I updated to asterisk 1.2.14 and add-ons 1.2.5 with no luck. I still don't see any update statement in the mysql.log when I change a password. I built a vmbox in the voicemail.conf file and can change that password just fine. Any suggestions? Thanks. JR ___ --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] windows mobile 5 softphone for square screen devices
Subject: Re: [asterisk-users] windows mobile 5 softphone for square screen devices Anton Krall a écrit : Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones? We are using PPCIAX. -- Daniel Hi, I found this from a google search. I have not tried it. https://www.ssldatas.com/globaliptel/(dv4ivf45q5vnz33azj1g4255)/_Pages/NoFrames/PageBuilder.aspx?content=52de526e3499426c875ed35f72ec935fhttps://www.ssldatas.com/globaliptel/%28dv4ivf45q5vnz33azj1g4255%29/_Pages/NoFrames/PageBuilder.aspx?content=52de526e3499426c875ed35f72ec935f You can also find the x-Lite - Soft SIP IP Phone 1.01 in this link below. I used this with PPC 2003 but I have not tried it with my new WM5.0: http://www.pdastreet.com/software/pdas/X-Lite-Soft-SIP-IP-2003-4-13-pdastreet-pdas.html Regards, Buki ___ --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] Erratic Snom MWI lights
Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Do you have an asterisk extension in your dialplan? See http://www.voip-info.org/wiki-Asterisk+phone+snom, especially the part about: Making the MWI work with ASTERISK Asterisk sends notifications on voicemail messages (if you configured the mailbox option in sip.conf. The messages are sent by default from [EMAIL PROTECTED], which can be modified using vmexten= in sip.conf. When pressing the MWI or Vmail soft button on the SNOM phones the phone calls this extension to connect to the voicemail application. If you haven't configured an extension named asterisk in the context of the phone, the MWI/VMail button will not work. Cheers, Stefano ___ --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] Erratic Snom MWI lights
I have not confirmed this independently, but I believe this is fixed if you disable the Show message light when a call is missed feature in the phone config. Alternatively, try pressing X to clear the missed call indication before pressing Retrieve Might work... Might not :) Steve On 1/17/07, J. Oquendo [EMAIL PROTECTED] wrote: Long story short... Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Useragent: snom360/6.5.2 Function: F_RETRIEVE [EMAIL PROTECTED] ~]# asterisk -rx show version Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on 2006-11-17 16:35:22 UTC [gateway] exten = 201,hint,SIP/201 exten = 201,1,Dial(SIP/201|20|tr) exten = 201,2,Voicemail([EMAIL PROTECTED]) exten = 201,3,Hangup -- ___ --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] FW: Realtime Voicemail Password Change Not Working
On 1/17/07, JR Richardson [EMAIL PROTECTED] wrote: I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, enter new password ok, re-enter new password ok, password has been changed There are no entries in the mysql.log setting the new password in the database. How can I isolate between asterisk, realtime driver, and mysql? I updated to asterisk 1.2.14 and add-ons 1.2.5 with no luck. I still don't see any update statement in the mysql.log when I change a password. I built a vmbox in the voicemail.conf file and can change that password just fine. Any suggestions? JR, I'm just pulling things out of the air here, but if realtime voicemail works like realtime users/peers, loading everything into memory from MySQL, then there would need to be some type or prune command to force the re-read of the voicemail table, this is asuming you change the password via MySQL and not on the handset. Maybe something like DBput would work to update astdb as well. Again just throwing out ideas... It sounds like you are using the handset to update the password. Is this correct? Regards, David ___ --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] Erratic Snom MWI lights
Steve Davies wrote:
I have not confirmed this independently, but I believe this is fixed
if you disable the Show message light when a call is missed feature
in the phone config. Alternatively, try pressing X to clear the
missed call indication before pressing Retrieve
Might work... Might not :)
Steve
Will try this, as for the other response: Tried them all. No dice
exten = asterisk,1,VoiceMailMain()
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
exten = asterisk,1,VoiceMailMain([EMAIL PROTECTED])
--
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net
The happiness of society is the end of government.
John Adams
smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [asterisk-users] FW: Realtime Voicemail Password Change Not Working
On 1/18/07, JR Richardson [EMAIL PROTECTED] wrote: I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, enter new password ok, re-enter new password ok, password has been changed There are no entries in the mysql.log setting the new password in the database. How can I isolate between asterisk, realtime driver, and mysql? I updated to asterisk 1.2.14 and add-ons 1.2.5 with no luck. I still don't see any update statement in the mysql.log when I change a password. I built a vmbox in the voicemail.conf file and can change that password just fine. Any suggestions? Thanks. JR Not sure about how you log in MySQL but using ODBC, in your odbcinst.ini or a similar file for Mysql, which keeps the settings for your db driver etc, you should be able to turn on logging. I can in odbcinst and it creates logs. The problem you have seems more like a permission problem however, the user you're using to log into the DB doesn't seem to have the permission to write to the table which keeps the user information OR the voicemail database itself. This problem becomes a bit trickier when your vm user table is actually a view of tables that hold subscriber/user information and is compounded by the fact if voicemails are being stored in a different db than where the sip/iax user information is being stored to derive sipusers and sippeers family values as then the user that asterisk is using to connect to the voicemail db will also need write permission in the db that stores user information. I dunno if any of that made sense but the password change works for me fine in 1.2.x as well as 1.4b3, haven't tested 1.4 Release yet. But in short, check pemissions for the user accessing the db(s) HTH \R ___ --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] Polycom IP601 - some hints working, not others?
Hi, I've just realised - the directory entries that have working Buddy watch are the first in sequence when the extensions are sorted into NAME order, which the phones do when saving their directory files. Looks like it could be a watch limit in that version of the firmware? Robert Jenkins. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Jenkins Sent: 16 January 2007 20:44 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Polycom IP601 - some hints working, not others? Hi, I've got an Asterisk setup including a TDM2400 for analog trunks extensions plus two IP501s an IP601 (all firmware 1.6.7 as supplied). The initial buddy / hint setup was fairly straightforward, but I have a strange problem in that some extensions don't show any status indication. Asterisk (V 1.2.13) CLI report for 'show hints' seems to indicate that the hints are set up correctly, but the phones are just not attempting to monitor certain extensions:- -= Registered Asterisk Dial Plan Hints =- 304: ZAP/4 State:Idle Watchers 0 303: ZAP/3 State:Idle Watchers 0 302: ZAP/2 State:Idle Watchers 0 301: ZAP/1 State:Idle Watchers 0 210: ZAP/16State:Idle Watchers 0 209: ZAP/15State:Idle Watchers 0 208: ZAP/14State:Idle Watchers 3 207: ZAP/13State:Idle Watchers 3 206: ZAP/12State:Idle Watchers 3 205: ZAP/11State:Idle Watchers 0 204: ZAP/10State:Idle Watchers 3 203: ZAP/9 State:Idle Watchers 0 202: SIP/202 State:Idle Watchers 3 201: SIP/201 State:Idle Watchers 3 200: SIP/200 State:Idle Watchers 3 The (mac)-directory.xml files have all the extensions in, in identical format, but the phones simply don't seem to be subscribing to certain 'buddys' to show the status. I've tried deleting directory entries at both an IP501 and the IP601 and re-creating them, with without rebooting, but with no effect. All entries have buddy watch enabled. The list of working / not working indications is consistent across reboots of both the phones and the Asterisk PC. Any ideas appreciated, Robert Jenkins. ___ --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 ___ --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] AbsoluteTimeout with canreinvite=yes
Is AbsoluteTimeout designed to work with canreinvite=yes? If not, are the any other options for disconnecting a call after a predefined duration when using canreinvite=yes? Thanks! David ___ --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] AbsoluteTimeout with canreinvite=yes
David Thomas wrote: Is AbsoluteTimeout designed to work with canreinvite=yes? If not, are the any other options for disconnecting a call after a predefined duration when using canreinvite=yes? Reinvites only reinvite the MEDIA, not the SIGNALING. It should work, but I've not tested it ___ --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 1.4 Hint is not detected the extensions status
Dear Friends and Supporters! I try to install the Asterisk 1.4, and I needs to activate the hint to for the call pickup feature. However, the hint is enabled and I can see the status of the extensions by run command show hints. It show the phones are Idle. However, it would NOT be able to detect the extensions if they are ringing or Inuse at all, that why it did not sent the ringnotify to the subscribed phone. When the phones are ringing or in talking, the show hints still show the phones are Idle. Any ideal? I would very appreciated for your help. Lan. ___ --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] Erratic Snom MWI lights
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
I use that one myself. Does the Snom attempt to dial asterisk when
you hit Retrieve? What error do you get? Sure it's in the right
context (I screw that up ALL the time)...
Stefano
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[asterisk-users] One way choppy sound
Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)==sip==(asterisk 1)iax2 trunk(asterisk 2) ===alaw==(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i don't understand why the sound is bad in only one way. Any sugestions to solve it more than welcome Thanks Yelson ___ --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] One way choppy sound
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 What kind of pstn interface are you using? For analog interfaces try adjusting txgain in zapata.conf. Yelson Vivas wrote: Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)==sip==(asterisk 1)iax2 trunk(asterisk 2)===alaw==(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i don't understand why the sound is bad in only one way. Any sugestions to solve it more than welcome Thanks Yelson -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFrm+f2QVs8jsa1mQRAjCvAKCxTfBey0nFtAIMr7RdQ5udimtjSQCfdCZE z+TSxAKsoyap2OXv4IQ1PMA= =4Ld/ -END PGP SIGNATURE- ___ --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] Callback/ringback
Richard Soderblom wrote: Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is busy or unavailable then Asterisk should inform the first user that the number they dialed is busy and hangup the call. Once the second caller is available again then Asterisk should initiate a call back to both the users and connect them. Any ideas on how to achieve this will be appreciated. Richard, That shouldn't be too difficult to do. I recently wrote an agi binary that does nag calling for me which I think is related to what you want to do except that I am doing more calling out of the system. Maybe deadagi could work? Here is it's use in a AEL macro I'm working on: http://www.datatrakpos.com/pos/datatalk/images/nagcall.htm The AGI (nagcall) simply takes some parameters and uses them to create a .call file. See: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out There is even an example on the page above that shows using the linux touch command to schedule the call to take place at a later time, although I have not successfully done this yet...still trying. The biggest difference is that you will need a way to monitor the called extension to trigger a call back to the original caller using maybe deadagi or .call files? http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+API I'm pretty new to asterisk myself so there may be (probably are) other ways to do this, but this is where I would start poking around. -- Warm Regards, Lee ___ --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] Network\Snom phone oddity
I have a client that has 5 Snom 320s. 4 work great, one does not. I upgrade the firmware to the latest (6.5.2) and the problem goes away, but then comes back a couple days later. There is a slight packet loss on the phone (about 1%), though there is no packet loss on any of the other phones. I determine the packet loss by the Linux command ping -f -c 1 192.168.2.10. Outgoing calls are junk, incoming calls are fine. (relatively speaking) The config from one phone to the next is the same except for account and voicemail settings. sip.conf is the same except for account. okay, the phone is bad, so I order a new one. This phone, however, is reporting 4% - 30% packet loss so every call is horrible just due to the lost packets (I'd assume). I install a new cable into a different port on the switch (same port as a working phone, with the working phone going into the same port as the old cable). Same results. Take this phone elsewhere. Packet loss continues. I even try different power supplies and handsets to find SOME sort of fault other than the obvious. I take the old phone back to my office and it works flawlessly, though my client uses the phone constantly all day whereas we only did approximately a half hour of testing. I take the new phone back to my office and it now has 0% packet loss. So, do I have two broken phones or is there something else wrong? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --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 1.4 and CDR
I have solved the problem, I have already install mysql-devel and then # cd asterisk-addons-1.4.0 # make distclean # ./configure # make # make install # make samples My Call Detail Records is running. 2007/1/17, Savoy, Kevin - Williston, ND [EMAIL PROTECTED]: I had the same issue. I needed to install #yum install mysql-devel. Once I did this the addons compiled the file fine. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Pablo Almido *Sent:* Wednesday, January 17, 2007 9:43 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Asterisk 1.4 and CDR Hi guys, I have recently installed a Asterisk Server with CDR Call Detail Records. I have installed it over a Asterisk 1.2 , but now It do not run . I have installed it with the following procedure: # yum install ncurses #yum install openh323-devel # yum install mysql-server # yum install mysql # yum install php-gd # yum install php-mysql # yum install mysqlclient10 # yum install zlib # yum install zlib-devel # yum install ncurses-devel Install perl support perl -MCPAN -e install DBD::mysql I compile /usr/src/asterisk-addons as follows: # ./configure # make clean # make install In the file /etc/asterisk/cdr_mysql.conf [global] hostname=localhost dbname=asteriskcdrdb table=cdr password=strongpass user=asterisk port=3306 userfield=1 In the File asterisk-stat define (WEBROOT, http://192.168.190.10/asterisk-stat/;); define (FSROOT, /var/www/html/asterisk-stat-v2/); define (LIBDIR, FSROOT.lib/); define (HOST, localhost); define (PORT, 3306); define (USER, asterisk); define (PASS, strongpass); define (DBNAME, asteriskcdrdb); define (DB_TYPE, mysql); // mysql or postgres define (DB_TABLENAME, cdr); When I compile asterisk-addons it pass very good, but I do not build the file *cdr_addon_mysql.so* Do you have similar problem ?Thanks for your response. Excuseme for my english, it is not my native language. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 1.4 and CDR
I have the same problem. Please reply to the list if you figure it out. I'll do the same. _ From: Pablo Almido [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 17, 2007 9:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4 and CDR Hi guys, I have recently installed a Asterisk Server with CDR Call Detail Records. I have installed it over a Asterisk 1.2 , but now It do not run . I have installed it with the following procedure: # yum install ncurses #yum install openh323-devel # yum install mysql-server # yum install mysql # yum install php-gd # yum install php-mysql # yum install mysqlclient10 # yum install zlib # yum install zlib-devel # yum install ncurses-devel Install perl support perl -MCPAN -e install DBD::mysql I compile /usr/src/asterisk-addons as follows: # ./configure # make clean # make install In the file /etc/asterisk/cdr_mysql.conf [global] hostname=localhost dbname=asteriskcdrdb table=cdr password=strongpass user=asterisk port=3306 userfield=1 In the File asterisk-stat define (WEBROOT, http://192.168.190.10/asterisk-stat/;); define (FSROOT, /var/www/html/asterisk-stat-v2/); define (LIBDIR, FSROOT.lib/); define (HOST, localhost); define (PORT, 3306); define (USER, asterisk); define (PASS, strongpass); define (DBNAME, asteriskcdrdb); define (DB_TYPE, mysql); // mysql or postgres define (DB_TABLENAME, cdr); When I compile asterisk-addons it pass very good, but I do not build the file cdr_addon_mysql.so Do you have similar problem ?Thanks for your response. Excuseme for my english, it is not my native language. ___ --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] Re: Realtime Voicemail Password Change Not Working
On 1/17/07, JR Richardson [EMAIL PROTECTED] wrote: I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, enter new password ok, re-enter new password ok, password has been changed There are no entries in the mysql.log setting the new password in the database. How can I isolate between asterisk, realtime driver, and mysql? I updated to asterisk 1.2.14 and add-ons 1.2.5 with no luck. I still don't see any update statement in the mysql.log when I change a password. I built a vmbox in the voicemail.conf file and can change that password just fine. Any suggestions? JR, I'm just pulling things out of the air here, but if realtime voicemail works like realtime users/peers, loading everything into memory from MySQL, then there would need to be some type or prune command to force the re-read of the voicemail table, this is asuming you change the password via MySQL and not on the handset. Maybe something like DBput would work to update astdb as well. Again just throwing out ideas... It sounds like you are using the handset to update the password. Is this correct? Yes, I'm using the handset to change the vm password, throught the vm advanced features. I can directly change the db as expected, just not through the vm application. Not sure about how you log in MySQL but using ODBC, in your odbcinst.ini or a similar file for Mysql, which keeps the settings for your db driver etc, you should be able to turn on logging. I can in odbcinst and it creates logs. The problem you have seems more like a permission problem however, the user you're using to log into the DB doesn't seem to have the permission to write to the table which keeps the user information OR the voicemail database itself. This problem becomes a bit trickier when your vm user table is actually a view of tables that hold subscriber/user information and is compounded by the fact if voicemails are being stored in a different db than where the sip/iax user information is being stored to derive sipusers and sippeers family values as then the user that asterisk is using to connect to the voicemail db will also need write permission in the db that stores user information. I use the same database for the sip, iax, exten and vm, different tables. When a sip device registers, asterisk writes to the database with updates to the sip table ipaddress, port and regseconds, so I don't think there is a write permissions issue from asterisk res_mysql to the mysql database. I thought of that also and changed the user to full access, but that didn't help. Mysql logs all database transactions in the /var/log/mysql.log file. I see all the query selects from the voicemail table and i see all the query updates to the sip table, but never see any query updates for the vmpasswd to the voicemail table. I would assume there would at least be errors if there was a permissions problem. I don't see where asterisk is trying to update the vmpassword through the realtime driver. How is your voicemail.conf file setup? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] Network\Snom phone oddity
Had this happen when I put a SNOM 360 On the Lan over Fiber. The Fiber transivers where stuck to 100tx and it was botching things. I put a 10base hub between the fiber and the phone and it worked. Disable the auto network config and I think that you can set the unplug to ignore and another setting logging into one. Phone Type: snom360-SIP MAC-Address:x IP-Address: 192.168.1.79 Kernel Version: snom360 linux 3.25 Application-Version:snom360-SIP 6.5.1 Rootfs-Version: snom360 jffs2 v3.36 Firmware-URL: http://192.168.20.1/snom/firmware/snom360-3.25-l.bin Production Information: Mac:00041323195B;Version:Standard;Hardware:snom360 (Revesion B);Lot:12 (June 2005) Thats right I had to update the Linux Kernel, the firmware and the filesystem three updates give it a try On 1/17/07, Mike Hammett [EMAIL PROTECTED] wrote: I have a client that has 5 Snom 320s. 4 work great, one does not. I upgrade the firmware to the latest (6.5.2) and the problem goes away, but then comes back a couple days later. There is a slight packet loss on the phone (about 1%), though there is no packet loss on any of the other phones. I determine the packet loss by the Linux command ping -f -c 1 192.168.2.10. Outgoing calls are junk, incoming calls are fine. (relatively speaking) The config from one phone to the next is the same except for account and voicemail settings. sip.conf is the same except for account. okay, the phone is bad, so I order a new one. This phone, however, is reporting 4% - 30% packet loss so every call is horrible just due to the lost packets (I'd assume). I install a new cable into a different port on the switch (same port as a working phone, with the working phone going into the same port as the old cable). Same results. Take this phone elsewhere. Packet loss continues. I even try different power supplies and handsets to find SOME sort of fault other than the obvious. I take the old phone back to my office and it works flawlessly, though my client uses the phone constantly all day whereas we only did approximately a half hour of testing. I take the new phone back to my office and it now has 0% packet loss. So, do I have two broken phones or is there something else wrong? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] I need to connect Asterisk to a Nortel Meridian phone plant
Hi. I had successful confiured my Asterix PBX, but now I need to connect it to a Nortel Meridian phone plant, I got a Digium T100P on the machine I got the Asterisk and I already connect it with the phone plant, I need to know what configuration do I have to do so the ip extensions can make calls to the extensions of my plant Nortel. And extensions from the Nortel plant can call ip extensions of Asterisk? I´m located in Colombia -- Gustavo Andrés Salazar Giraldo ___ --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] Network\Snom phone oddity
On the voip-info.org wiki there are good tips to get snoms to play nice on lans. I personally have experienced wierdness using particular switches (cheap ones). also note that snom now has an auto-update subscription URL in their support wiki, if you use the URL it makes updating a 4.X to 6.X with the new FS, linux, etc not quite so brutal. ___ --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] Erratic Snom MWI lights
I use that one myself. Does the Snom attempt to dial asterisk when
you hit Retrieve? What error do you get? Sure it's in the right
context (I screw that up ALL the time)...
Sometimes it's asterisk, sometimes it's unknown sometimes, it's
Unknown so:
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
exten = Unknown,1,VoicemailMain(${CALLERIDNUM})
exten = unknown,1,VoicemailMain(${CALLERIDNUM})
This behavior varies seems like according to firmware rev. I just put these
lines in the default context (I run a modified version of the AMP dialplan,
so the context I use is [from-internal]
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Re: [asterisk-users] I need to connect Asterisk to a Nortel Meridian phone plant
On Wednesday 17 January 2007 3:35 pm, Gustavo Andrés Salazar Giraldo wrote: Hi. I had successful confiured my Asterix PBX, but now I need to connect it to a Nortel Meridian phone plant, I got a Digium T100P on the machine I got the Asterisk and I already connect it with the phone plant, I need to know what configuration do I have to do so the ip extensions can make calls to the extensions of my plant Nortel. And extensions from the Nortel plant can call ip extensions of Asterisk? I´m located in Colombia I have an Asterisk machine sitting between a Norstar MICS with a PRI trunk, and the telephone company's PRI. Being in Columbia I am guessing that you are using E1/PRI. What is your connection to the phone network? -A. ___ --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] Queue and Interface time out
Hmm, the use of autopause in queues.conf introduces a new issue. When a queue member is on a call, the queue continues to try to send calls to the member's interface. Getting the 'Busy Here' response from the SIP device causes the caller to continue holding. The new issue is that autopause appears to pause the member interface even when they're on another call. Am I missing something or is this the expected behavior? I didn't expect the Queue application to try member interfaces that are busy. Thanks, James James Fromm wrote: NICE! That did the trick. Thanks! Julian Lyndon-Smith wrote: try autopause in queues.conf James Fromm wrote: We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James ___ --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 ___ --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 ___ --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 ___ --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] Unknown warning messages
Hi folks, When my Sipura 2k is registered to Asterisk, I get some peculiar error messages repeated in the logs every 30 seconds. I've put a snippet up in http://pastebin.co.uk/9067 for you to see. I don't have any complicated setups. Just 2 sip.conf entries for the Sipura, and 2 more for softphones. They can all register just fine, and I can make a call from the Sipura to a test extension. If anyone is able to provide some insight into this, it would be greatly appreciated. Regards, Kyle ___ --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] Erratic Snom MWI lights
On Wednesday 17 January 2007 3:43 pm, Colin Anderson wrote:
Sometimes it's asterisk, sometimes it's unknown sometimes, it's
Unknown so:
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
exten = Unknown,1,VoicemailMain(${CALLERIDNUM})
exten = unknown,1,VoicemailMain(${CALLERIDNUM})
This behavior varies seems like according to firmware rev. I just put these
lines in the default context (I run a modified version of the AMP dialplan,
so the context I use is [from-internal]
This is just a plain old bad idea, especially if your caller id is not
consistently and religiously set up. It seems like you're trusting the
phones too much.
Personally, my voicemail stuff looks a little like this:
exten = _*98XXX,1,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
exten = *97,1,GotoIf($[${LEN(${CALLERID(num)})} == 3]?good)
exten = *97,n,Set(VMBOX=)
exten = *97,n(done),VoiceMailMain([EMAIL PROTECTED])
and then the standard extension:
; if VOICEMAILTO is set, send voicemail to there instead of ${ARG1}
[macro-stdexten]
exten = s,1,GotoIf($[ ${VOICEMAILTO} != ]?vmto)
exten = s,n,Set(VOICEMAILTO=${ARG1})
exten = s,n(vmto),Dial(${ARG2},20,g)
exten = s,n,GotoIf($[${DIALSTATUS} = BUSY]?busy)
exten = s,n,GotoIf($[${DIALSTATUS} = CONGESTION]?busy)
exten = s,n,GotoIf($[${DIALSTATUS} = ANSWER]?done)
exten = s,n,Voicemail([EMAIL PROTECTED],su)
exten = s,n,Goto(done)
exten = s,n(busy),Voicemail([EMAIL PROTECTED],sb)
exten = s,n(done),Hangup
The phones (polycom in my case) are set that vmcallback is *97, and *98XXX is
used more for administrators and DISA-style applications.
-A.
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Re: [asterisk-users] force ulaw passthrough if call from modem extension?
On 1/17/07, Victor Perez [EMAIL PROTECTED] wrote: Tried that, it didn't work but maybe I didn't configure it right. Anyways how can I route all outgoing calls from that specific extension to use that trunk? Put that extension in a different context. ___ --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] I need to connect Asterisk to a Nortel Meridian phone plant
Yes, i am using E1/PRI Thanks, for your help. On 1/17/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Wednesday 17 January 2007 3:35 pm, Gustavo Andrés Salazar Giraldo wrote: Hi. I had successful confiured my Asterix PBX, but now I need to connect it to a Nortel Meridian phone plant, I got a Digium T100P on the machine I got the Asterisk and I already connect it with the phone plant, I need to know what configuration do I have to do so the ip extensions can make calls to the extensions of my plant Nortel. And extensions from the Nortel plant can call ip extensions of Asterisk? I´m located in Colombia I have an Asterisk machine sitting between a Norstar MICS with a PRI trunk, and the telephone company's PRI. Being in Columbia I am guessing that you are using E1/PRI. What is your connection to the phone network? -A. ___ --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 -- Gustavo Andrés Salazar Giraldo Ingeniero Telemático - Universidad Icesi Celular: 310 4593225 ___ --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] Polycom IP601 - some hints working, not others?
I've just realised - the directory entries that have working Buddy watch are the first in sequence when the extensions are sorted into NAME order, which the phones do when saving their directory files. Looks like it could be a watch limit in that version of the firmware? Could it be that it's only watching the one's it has available lines to give a buddy line to? Example, I have my IP601 (6 line phone) setup with 3 lines worth of my extension (so I can more easily handle three calls at once). That leaves the other three for the buddies I use most so I can direct dial and monitor them with the in-use light. I also have the hint application in my extentions.conf dialplan, I'm assuming you have that covered. -Kenneth ___ --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] STUN in Asterisk 1.4
Browsing through the developers documentation and 1.4 source, I see references to STUN in the code and documentation. Does 1.4 have support for STUN, if so how is it configured? Regards, David ___ --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] windows mobile 5 softphone for square screen devices
Well Guys.. I just bought the X-PDA one and indeed it has 240x240 support, costs around 29 USD and very well worth it, works great on an iPAQ 6945 via wireless and using my BT headset all sound goes to the headset and not the speaker, which is great and solves the eternal problem of having to listen to your call thru the speaker (not the phone speaker but the hands free one) AK |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Timothy Parez |Sent: Wednesday, January 17, 2007 9:40 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] windows mobile 5 softphone for square screen devices | |Hi, | |I've been looking for a good SIP application for Windows Mobile for ages. |I found speaQ, but it has the same problem as any other softphone for |Windows Mobile. | |You see, it uses the speaker to output the conversation instead of the |phone speaker, |you know the one that is used when you make a normal phone call with |your WM Mobile PDA/Smartphone. |At first I was asking myself if every SIP client developer out there is |down right stupid but |in the end I found out this is actually Microsoft blocking access to |that phone speaker. |The claim that allowing the developers to access it would allow for |invasion of privacy (like recording phone calls). |So unless someone can work around this, softphones for WM will remain |quite useless. | |Timothy. | |Anton Krall wrote: | Guys, anybody has seen or is using some kind of softphone on any square | screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they | do work on Wm5 but they are designed for standard screens, anybody using | anything on square ones? | | | | ___ | --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 | | | |___ |--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 ___ --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] Queue and Interface time out
James Fromm wrote: Hmm, the use of autopause in queues.conf introduces a new issue. When a queue member is on a call, the queue continues to try to send calls to the member's interface. Getting the 'Busy Here' response from the SIP device causes the caller to continue holding. The new issue is that autopause appears to pause the member interface even when they're on another call. Am I missing something or is this the expected behavior? queues.conf: ; Autopause will pause a queue member if they fail to answer a call ; ;autopause=yes I didn't expect the Queue application to try member interfaces that are busy. queues.conf: ; If you want the queue to avoid sending calls to members whose devices are ; known to be 'in use' (via the channel driver supporting that device state) ; uncomment this option. (Note: only the SIP channel driver currently is able ; to report 'in use'.) ; ; ringinuse = no Julian Thanks, James James Fromm wrote: NICE! That did the trick. Thanks! Julian Lyndon-Smith wrote: try autopause in queues.conf ___ --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 Legacy PBX integration and fail-over question,
Hi All, I have a (legacy) Praxton PBX, it has a PRI T1 input card and 64 analog extensions through 4 amphenol connectors. We receive 12 voice channels (other 12 are idle) and have 100 DIDs. No caller ID thru PRI though. The Praxton box is amazing in terms of configuration and flexibility but has no VoIP support and the company went poof and it is no longer supported, nor spare parts are available, and it's giving me a hiccup at least once a week. I already have good experience with Asterisk installs using TDM cards, but never used TE T1s before. Saying that, I am researching for a full replacement, (most options are adding up to $13k+) so, why not also considering a cheaper fail-over solution that only implies purchasing one TDM and one TE2 Is this scenario possible? 1./ T1 line from the telco with signaling changed to receive caller IDs (telco tells me ok) 2./ Telco T1 enters Asterisk Box through a dual TE2 card 3./ Box forwards calls to the legacy PBX trough second port on dual TE2 with companion Caller ID info (obviously connected to the legacy PBX) if the legacy box is down, then the calls get routed trough a TDM4 to an analog phone, or even better, the * Box runs some test on the T1 to the legacy box and rings all four phones on the TDM4 when the legacy box is down, and of course the Asterisk box can provide me all the connectivity and flexibility Asterisk is capable off, with still 12 free channels in all T1 ports Thank you very much in advance for the enlightenment you'll bring Andres ___ --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] Queue and Interface time out
DoH! I missed that ringinuse. Thanks! Julian Lyndon-Smith wrote: James Fromm wrote: Hmm, the use of autopause in queues.conf introduces a new issue. When a queue member is on a call, the queue continues to try to send calls to the member's interface. Getting the 'Busy Here' response from the SIP device causes the caller to continue holding. The new issue is that autopause appears to pause the member interface even when they're on another call. Am I missing something or is this the expected behavior? queues.conf: ; Autopause will pause a queue member if they fail to answer a call ; ;autopause=yes I didn't expect the Queue application to try member interfaces that are busy. queues.conf: ; If you want the queue to avoid sending calls to members whose devices are ; known to be 'in use' (via the channel driver supporting that device state) ; uncomment this option. (Note: only the SIP channel driver currently is able ; to report 'in use'.) ; ; ringinuse = no Julian Thanks, James James Fromm wrote: NICE! That did the trick. Thanks! Julian Lyndon-Smith wrote: try autopause in queues.conf ___ --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 ___ --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] Rt db lookup
Okay. That doesn't help. What forces * to look at the DB rather than waiting on a registration ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Monday, January 15, 2007 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Rt db lookup On 1/15/07, Tim Connolly [EMAIL PROTECTED] wrote: Which command effects whether or not the * server will lookup a peer from the db even though the phone isn't registered locally? I have several * servers but I want any server to be able to lookup and send a call to phones registered on another server (SIP cluster?). You may want to look at DUNDi for this. http://www.dundi.info/ regards David ___ --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 ___ --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] Rt db lookup
On 1/17/07, Tim Connolly [EMAIL PROTECTED] wrote: Okay. That doesn't help. What forces * to look at the DB rather than waiting on a registration ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Monday, January 15, 2007 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Rt db lookup On 1/15/07, Tim Connolly [EMAIL PROTECTED] wrote: Which command effects whether or not the * server will lookup a peer from the db even though the phone isn't registered locally? I have several * servers but I want any server to be able to lookup and send a call to phones registered on another server (SIP cluster?). You may want to look at DUNDi for this. http://www.dundi.info/ regards David To my knowledge, the only two ways to do this is... 1.) To create a SIP or IAX trunk between each box that needs to communicate then add the login to your dialplan in extensions.conf to use those trunks when the call cannot be completed locally. 2.) To create a SIP or IAX trunk between each box that needs to communicate then configure DUNDi to handle the extension location. As far as registration and Realtime is concerned... have a look at the rtcachefriends option in sip.conf iax.conf. Hope this helps. - David ___ --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] Rt db lookup
On 1/17/07, David Thomas [EMAIL PROTECTED] wrote: On 1/17/07, Tim Connolly [EMAIL PROTECTED] wrote: Okay. That doesn't help. What forces * to look at the DB rather than waiting on a registration ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Monday, January 15, 2007 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Rt db lookup On 1/15/07, Tim Connolly [EMAIL PROTECTED] wrote: Which command effects whether or not the * server will lookup a peer from the db even though the phone isn't registered locally? I have several * servers but I want any server to be able to lookup and send a call to phones registered on another server (SIP cluster?). You may want to look at DUNDi for this. http://www.dundi.info/ regards David To my knowledge, the only two ways to do this is... 1.) To create a SIP or IAX trunk between each box that needs to communicate then add the login to your dialplan in extensions.conf to use those trunks when the call cannot be completed locally. 2.) To create a SIP or IAX trunk between each box that needs to communicate then configure DUNDi to handle the extension location. As far as registration and Realtime is concerned... have a look at the rtcachefriends option in sip.conf iax.conf. Hope this helps. - David What I meant to say on # 1 is add the logic to your dialplan in extensions.conf - David ___ --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] Hospitals using Asterisk?
Hello, The IT folks at a hospital in New Zealand have approached us about deploying Asterisk, but they would like to talk to people at other hospitals that have already done this. If anyone works at a hospital that has deployed Asterisk, or deployed Asterisk at a hospital would you please get in touch with me? Either via email, or I'm currently at linux.conf.au in Sydney if you're here. Thanks! -- Andrew Ruthven, Wellington, New Zealand At work: [EMAIL PROTECTED] At home: [EMAIL PROTECTED] GPG fpr: 34CA 12A3 C6F8 B156 72C2 D0D7 D286 CE0C 0C62 B791 signature.asc Description: This is a digitally signed message part ___ --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] Queue and Interface time out
I guess I'm missing something else. 'ringinuse = no' doesn't change anything. While on a call, the queue still sends another call and proceeds to set the member paused after receiving 'Busy Here' back from the SIP device. My queues.conf is: [general] persistentmembers = no [customerservice] persistentmembers = no musiconhold = default reportholdtime = no strategy = leastrecent timeout = 20 retry = 5 wrapuptime = 30 ;allow agents 30 seconds to wrap up work maxlen = 0 ;unlimited callers on hold servicelevel = 60 ;calls must be answered within 60 seconds announce-holdtime = no autopause = yes ringinuse = no joinempty = yes leavewhenempty = no I'm I missing something obvious? Thanks, James James Fromm wrote: DoH! I missed that ringinuse. Thanks! Julian Lyndon-Smith wrote: James Fromm wrote: Hmm, the use of autopause in queues.conf introduces a new issue. When a queue member is on a call, the queue continues to try to send calls to the member's interface. Getting the 'Busy Here' response from the SIP device causes the caller to continue holding. The new issue is that autopause appears to pause the member interface even when they're on another call. Am I missing something or is this the expected behavior? queues.conf: ; Autopause will pause a queue member if they fail to answer a call ; ;autopause=yes I didn't expect the Queue application to try member interfaces that are busy. queues.conf: ; If you want the queue to avoid sending calls to members whose devices are ; known to be 'in use' (via the channel driver supporting that device state) ; uncomment this option. (Note: only the SIP channel driver currently is able ; to report 'in use'.) ; ; ringinuse = no Julian Thanks, James James Fromm wrote: NICE! That did the trick. Thanks! Julian Lyndon-Smith wrote: try autopause in queues.conf ___ --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 ___ --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 ___ --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] Dtmf tones and SIP
I aaume you are calling in on a PSTN line? If so what fxo are you using with Asterisk. Doug On Wed, 17 Jan 2007, Giuffredi wrote: Hi list, I tried to use DISA in order to get the line when I call with my mobile phone but the system doesn't recognise my DTMF tones when I call to a SIP trunk. Everything is working Ok if I use a ZAP Trunks. I tried to google to find a solution but I wasn't able to find any. Any idea? I'm using trixbox 2 with 4 SIP trunks and a Zaptel TDM 400 card. Bye Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --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] Re: Realtime Voicemail Password Change Not Working
On 1/18/07, JR Richardson [EMAIL PROTECTED] wrote: I use the same database for the sip, iax, exten and vm, different tables. When a sip device registers, asterisk writes to the database with updates to the sip table ipaddress, port and regseconds, so I don't think there is a write permissions issue from asterisk res_mysql to the mysql database. I thought of that also and changed the user to full access, but that didn't help. Mysql logs all database transactions in the /var/log/mysql.log file. I see all the query selects from the voicemail table and i see all the query updates to the sip table, but never see any query updates for the vmpasswd to the voicemail table. I would assume there would at least be errors if there was a permissions problem. I don't see where asterisk is trying to update the vmpassword through the realtime driver. How is your voicemail.conf file setup? Thanks. JR Interesting, well if you're seeing the other selects in the mysql.log then this update not showing up is bizarre. It would also mean that permissions are irrelevant if doesn't even attempt to change the password, as you'd rightly pointed out as well. I just tested it again and this is what I see in the odbc sql.log SQL = [UPDATE vmusers SET password=? WHERE uniqueid=?][length = 46 (SQL_NTS)] So it definately spits out something but my setup is considerably different to yours. I am using ODBC - FreeTDS - MS SQL Server for starters. There's nothing out of the oridinary in my voicemail.conf. What I do remember is some conversation sometime about the file locking fix that was put in or was being talked about regards to people using static files and multiple people trying to change their passwords. Just checking if you've compiled (*) clean without any mods to the code etc. I mean I have made mods to app_voicemail.c but nothing that affects passwords. Just for giggles, have you tried doing realtime update voicemail mailbox 1234 password 2345 ? I know you said that your db updates for regsecs, ip address etc is working but try specifically writing to your voicemail table and see if you are able to manually update the password. At least that way you can just focus on seeing why the password code is not being triggered in the (*) code when using MySQL. Sorry I cna't think of anything else to suggest at the moment. HTH \R ___ --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 + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] 2 Questions: Answer with music don't work and Voicemail direct access ?
Hi Stefan,
Thanks for your answer, but it's a error of me in cut, the goto are good:
[Cal-In]
exten = _81120,1,Goto(C-Internal,100,1)
exten = _81121,1,Goto(C-Internal,200,1)
[C-Internal]
exten = 100,1,Ringing
exten = 100,2,Wait,1
exten = 100,3,Answer
exten = 100,4,Dial(SIP/220SIP/221,30)
exten = 100,5,Hangup
exten = 200,1,Ringing
exten = 200,2,Wait,1
exten = 200,3,Answer
exten = 200,4,Dial(SIP/221,25,tm)
exten = 200,5,Hangup
;=)
Stefan Wintermeyer a écrit :
Hi,
Am 17.01.2007 um 15:07 schrieb Noc Phibee:
Problems with Answer+Music
my extension:
[Cal-In]
exten = _81120,1,Goto(C-Internal,100,1)
exten = _81121,1,Goto(C-Internal,200,1)
[C-Phibee]
exten = 100,1,Ringing
exten = 100,2,Wait,1
exten = 100,3,Answer
exten = 100,4,Dial(SIP/201SIP/200,30)
exten = 100,5,Hangup
exten = 200,1,Ringing
exten = 200,2,Wait,1
exten = 200,3,Answer
exten = 200,4,Dial(SIP/200,25,tm)
exten = 200,5,Hangup
With this extension, when a incoming call are received :
If my customer have call 081120, that's answer and Ring
If my customer have call 081121, he have a answer, he have a
music
I don't know why the 081120 don't have the music for wait that i
am answer ...
I guess you simply did a mistake in the Goto. It points to the
C-Internal context but you want to jump to C-Phibee.
It's possible to put into the extension, for access to the VoiceMail,
the extension of the caller ?
exten = 500,1,VoiceMailMain(@Home)
Actually, when i call the 500, he want know my mailbox ID and after
password ...
if i call with the post 200, it's possible to access direclty at the
password ?
Yes:
exten = 500,1,VoiceMailMain(${CALLERID(num)[EMAIL PROTECTED])
But I am not sure if you really want to use @Home here. But that
depends on you voicemail.conf
BTW: With exten = 500,1,VoiceMailMain(${CALLERID(num)[EMAIL PROTECTED],s) you
can even skip the password question.
Stefan
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
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Re: [asterisk-users] DND - message
I had been wondering the same thing, I haven't really found any useful information. I use: exten = 123,1,Dial(SIP/123) exten = 123,2,Voicemail(u123) exten = 123,102,Voicemail(b123) If you set DND on the SIP phone usually it sends 486 busy here and jumps to 102. If you reject the call it usually sends 603 decline. Is there any way to get this value into the dialplan? I know you can avoid the on the phone message when the phone is not registered by using regexten. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback/ringback
Enclosed bellow is the fragment from extenstions.conf which does two things:
*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.
Regards, __Yehavi:
; regular local extensions:
; The flow is: If not available or no answer send to mailbox if exists,
; send busy if no mailbox. Same for busy.
; We try to avoid the n+101 rule whenever possible, but it is not always
; possible as HasVoiceMailbox() does only n+101 jump.
exten = _999XX,1,Set(_To=${EXTEN}) ; Save the original extension dialled.
exten = _999XX,n,Set(_From=${CALLERID(num)}) ; Save the caller.
; Save the caller number at the called extension for *42 usage.
exten = _999XX,n,Set(DB(${To}/LastCaller)=${From})
; Where we called for *41
exten = _999XX,n,Set(DB(${From}/LastCalled)=${To})
; Now dial the extension.
exten = _999XX,n,Dial(SIP/${EXTEN},20,) ; Dial the phone for 20 seconds.
; No answer or busy
exten = _999XX,n,GoTo(s-${DIALSTATUS},1) ; Jump according to the failure
mode
exten = _999XX,n,Hangup() ; Just to be sure...
; No answer:
exten = s-NOANSWER,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-NOANSWER,n,Busy(); No maibox = play busy.
exten = s-NOANSWER,102,VoiceMail(u${To}) ; Has mailbox - send the call to there
; Busy:
exten = s-BUSY,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-BUSY,n,Busy(); No maibox = play busy.
exten = s-BUSY,102,VoiceMail(b${To}) ; Has mailbox - send the call to there
; Unavailable channel - act as busy:
exten = s-CHANUNAVAIL,1,Goto(s-BUSY,1);
; Called here when the call is successfull and the user hanged the phone.
; Check whether the user has a waiting callback queued on him/her
exten = h,1,NoOp(${From} ${To} ${EXTEN})
exten = h,2,Set(tmp=${DB(${From}/CallBack)}) ; Get who is waiting for us
exten = h,3,NoOp(${From} ${tmp})
exten = h,4,GotoIf($[ ${tmp} ]?5:103) ; Anyone waiting for us?
exten = h,5,DBdel(${From}/CallBack); And delete it...
; Create the callfile and then move it to the spool directory to make the call.
exten = h,6,System(echo Channel: SIP/${tmp} /tmp/test.tmp${To})
exten = h,7,System(echo WaitTime: 20 /tmp/test.tmp${To})
exten = h,8,System(echo Extension: ${From} /tmp/test.tmp${To})
exten = h,9,System(echo CallerID: Callback \\\${tmp}\\\
/tmp/test.tmp${To})
exten = h,10,System(mv /tmp/test.tmp${To} /var/spool/asterisk/outgoing/)
exten = h,103,NoOp(Nothing to call)
; *42: Get the last number who called us, say it and call it.
exten = *42,1,Set(tmp=${DB(${CALLERID(num)}/LastCaller})
exten = *42,n,SayDigits(${tmp})
exten = *42,n,Goto(${tmp},1)
; *41: Camp on the last extension dialled
exten = *41,1,Set(tmp=${DB(${CALLERID(num)}/LastCalled)})
exten = *41,n,SayDigits(${tmp})
; Save it so when the other side hangs it will see it and dial us.
exten = *41,n,Set(DB(${tmp}/CallBack)=${CALLERID(num)})
exten = *41,n,Hangup()
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[asterisk-users] help. newbie asterisk installation problem.
Hello friends, I am trying to install asterisk 1.4.0 . I am configuring it as follows:- ./configure --prefix=/home/vivek/downloads/install/asterisk/ But still while running 'make install', it tries to install it in /var/lib/asterisk/ and stops because of failing permissions. I have provided it a prefix, But it doesn't install it there. Can anybody tell me the solution for this. I dont want to install it in the default directories. I want it to be in my home directory. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon Electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DND - message
Is there any reason why you could not do this?
exten = 123,1,Dial(SIP/123)
exten = 123,n,Goto(s-${DIALSTATUS},1)
exten = 123,n,HangUp
exten = s-NOANSWER,1,Voicemail(u123)
exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,Voicemail(b123)
exten = s-BUSY,2,Hangup
Or you could have pr-recorded messages in there before going to
Voicemail like
exten = s-BUSY,1,Backgroud(busy-call-back-ltr) ; If you had this sound
file.
.
.
.
.
exten = s-BUSY,n,Voicemail(b123)
exten = s-BUSY,n,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: Thursday, 18 January 2007 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DND - message
I had been wondering the same thing, I haven't really found any useful
information.
I use:
exten = 123,1,Dial(SIP/123)
exten = 123,2,Voicemail(u123)
exten = 123,102,Voicemail(b123)
If you set DND on the SIP phone usually it sends 486 busy here and
jumps to 102. If you reject the call it usually sends 603 decline. Is
there any way to get this value into the dialplan?
I know you can avoid the on the phone message when the phone is not
registered by using regexten.
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[asterisk-users] function call out of AGI script
Hi everyone, Is it possible to call an asterisk function out an AGI script? How do I do this? Thank you, Thomas ___ --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] sangoma a102d + Asterisk 1.2.14 ... bridging together 2 call legs on same PRI?
Hi all, Are there any issues to be concerned about when calls come in from PSTN to a PRI card and are forwarded back out the same PRI card? Anything different have to be enabled in zaptel.conf or zapata.conf or the Sangoma configs to make this work? What about using .call files that join two ZAP channels? Channel: ZAP/1/4081234567 MaxRetries: 0 RetryTime: 60 WaitTime: 60 Application: Dial Data: ZAP/1/4083456789 ___ --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] disable external-outgoing calls per extension
Hi all, Is there a simple way to disable external outgoing calls on the basis of calling extension ? In other words I would like to have two different groups of internal extensions, one enabled to place calls on the external PSTN, the other only enabled to place internal calls or receiving external calls. We take the external line using leading zero in dialed numbers. We are using FreePBX to manage Asterisk. Thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --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] TDM2400 Hardware Echo Cancel (Adam Sharples)
Hi Adam, I have the same problem. Are you sure is an echo canceller problem? Following advices from this list I discovered that I had an IRQ shared. Untill now I didn't try the new setup but I really hope that this was the problem. If you manage to solve the problem in any way please give me advise. Ciao You can make sure the card is sitting on it's own IRQ - use the command cat /proc/interrupts You can also check that the card isn't losing interrupts by running the zttest proram: /sbin/zttest There's more on interrupts here: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting It's aimes at the TDM400P card, but I'd be surprised if the 2400P is that much different (but someone please correct me if it is!) lspci -vb ___ --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] Error on answer a SIP 401 message
I'm a voip service provider and i'm setting up a asterisk box to register around 100 lines from my central softswitch. This asterisk box will be placed inside a customer and has a digium card to be interconected with customer's pabx. My problem is that when asterisk send register message, my softswitch return with sip 401 and asterisk should send a register message with Authorization in header. Only after 10, or 15 or 30 or 43 or 2 messages 401 asterisk start to send Authorization in header. This is a random time, don't follow any rule. I had something vaguely similar. Asterisk was replying on the wrong interface/network card. Might be worth checking. Cameron ___ --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] Using the SIPAddHeader Application
My *guess* is that the semicolon is being interpreted as a commend marker, so you might need to escape it with a '\'. I had problems with this, however, when using it via the [EMAIL PROTECTED] management interface, because the '\' is stripped on display and so lost if you view and save a working config file - I never did find a solution to this. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Hecker Sent: 17 January 2007 07:43 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Using the SIPAddHeader Application Hi, I'm trying to use the SIPAddHeader application to add a header containing to semicolon separated strings like this: exten = 12, 1, SIPAddHeader(X-TestHeader:a=test1;b=test2) But in the resulting INVITE message only the first part (X-TestHeader:a=test1) is added. Setting into quotation mark doesn't change anything. exten = 12, 1, SIPAddHeader(X-TestHeader:a=test1;b=test2) Do you have an idea how to achieve it? Thank you, Thomas ___ --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] Using the SIPAddHeader Application
Ok, this works pretty fine! Thank you very much! On 17/01/07, Steve Langstaff [EMAIL PROTECTED] wrote: My *guess* is that the semicolon is being interpreted as a commend marker, so you might need to escape it with a '\'. I had problems with this, however, when using it via the [EMAIL PROTECTED] management interface, because the '\' is stripped on display and so lost if you view and save a working config file - I never did find a solution to this. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Thomas Hecker *Sent:* 17 January 2007 07:43 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Using the SIPAddHeader Application Hi, I'm trying to use the SIPAddHeader application to add a header containing to semicolon separated strings like this: exten = 12, 1, SIPAddHeader(X-TestHeader:a=test1;b=test2) But in the resulting INVITE message only the first part (X-TestHeader:a=test1) is added. Setting into quotation mark doesn't change anything. exten = 12, 1, SIPAddHeader(X-TestHeader:a=test1;b=test2) Do you have an idea how to achieve it? Thank you, Thomas ___ --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 ___ --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] Question about FXO/FXS device.
Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about SPA3102. What you guys think about it. Is ok, is working with asterisk, can i use it like voip peer. Thank you for your advice. Jonson. ___ --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] Echo...
should have sent this to the list, Gordon how are you getting on with BT? - Original Message - From: Gordon Henderson [EMAIL PROTECTED] To: Wireless [EMAIL PROTECTED] Sent: Friday, January 12, 2007 10:45 PM Subject: Re: [asterisk-users] Echo... On Fri, 12 Jan 2007, Wireless wrote: http://www.voip-info.org/wiki-Asterisk+and+UK+Caller+ID see comments at the bottom Yup. Been through that, but I'll go back try the gains again. I got the Sangoma A200 from MyPhoneCall they seem to be the only place selling them in Blighty? would be interested if there is a cheaper source of course ;) Thats who I found (just using google though!) Gordon when you use the agressive canceller do you find it clips the speach? and what speed machine are you using? I'm using a P3 450mmx (using mmx in Zaptel) 256mb RAM maybe I need a faster machine to kill the Echo? The sound seems OK to me - certianly my wife's not complaining, neither are any of my customers (so far - apart from the one with the weird echo issue!) I'm using Mini ITX boards (VIA EPIA CN1000 boards) with a fanless 1GHz processor. I have to compile Asterisk for a i586 as they are lacking some MMX instructions, but are otherwise OK. My (home/office) test box is a 533MHz processor (same type, older version) Eric (ManxPower) can you give us an example of the EC you get from Ebay - are these something that could be integrated with FXO ports on an Asterisk box? I'd need to have a look, but I'm not sure I want to do down this route if I think I can get BT to fix it first! Incidentally, I saw your web site ... I built up 3 differenet community broadband systems using WiFi once upon a time. Intersting stuff but it never paid for itself in the end... Cheers, Gordon Thanks Harvey - Original Message - From: Gordon Henderson [EMAIL PROTECTED] To: Wireless [EMAIL PROTECTED] Sent: Friday, January 12, 2007 2:55 PM Subject: Re: [asterisk-users] Echo... On Fri, 12 Jan 2007, Wireless wrote: Gordon, if you can afford it go for a Sangoma A200 with hardware echo cancellation built in (it is my next step), the Sangoma support is excellent. I had a problem with CLID so I let Sangoma SSH into my Trixbox and they confirmed that my config was fine and suggested adjusting the gain - which fixed my intermittent CLID. May I ask who you are buying it from in the UK? My home/office system passes the WT! And I have a dozen or so boxes out in the field, but only this one which has the echo issue - however, I may be able to justify the additional cost to them if it makes it work. Intersting to hear about the callerId issues - I have issues too - I'll try the gain settings next! Thanks, Gordon -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --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] prompt for send a message not played in VM main, HOWTO resolve
On 1/17/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: All, Just came across the prompt #3 from inside the top menu of VM in latest stable. Allison does not announce the prompt, but if you know it is there, you can press 3 successfully follow the prompts from there to send your message to other users on the system. But, of course, obviously, I am asking: how do I resolve the situation whereby the users are not hearing this prompt? (since most nearly all users will never know that this is here) (I sure hope my googling didnt miss this one) Thanks very much. Most appreciated. Jason Sjobeck Jason, I dunno if I understand your question properly. Did you not want the prompt to play or did you want the prompt to play? If it's the latter, then AFAIK, this has to do with the setting in your voicemail.conf file which allows users to send messages to other users, it's sendvoicemail=yes, if you turn this on, you'll hear the prompt. If it's set to no then you won't hear the prompt to allow users to send msgs to other users. If that';s what you were asking then your googling did miss it :P HTH \R ___ --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] Re: [asterisk-dev] Question about FXO/FXS device.
Okay, i'll move my discuss to asterisk-users. Thank you. On 1/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jan 17, 2007 at 04:39:03PM +0800, 黄宗宁 wrote: Jonson Player wrote: Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about SPA3102. What you guys thik about it. Is ok, is working with asterisk, can i use it like voip peer. Thank you for your advice. Generally yes. This type of question should be asked on asterisk-users . This is a list for the development of Asterisk (ugly code stuff). Please follow-up there. Why done you buy FXO/FXS device from China.It is cheaper than other ones and more compact with asterisk. This belongs on either asterisk-users or asterisk-biz (if you happen to promote your own product). BTW: sorry for the messed encoding. I guess that sending a message in UTF-8 would have been safer. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --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] SPA-941 (and others ) Transmit Sound Quality
Hi guys, I got also problems with SPA 941 and 942, pour sound (a kind of click noise) that when i set volume sound lower almost can't notice, but still exists. I also notice on SIP to SIP calls , echo that could only be justified by Handsets hardware quality. When i make calls using Xlite with Plantronics DSP 400 USB micro and headset everything works like a charm. I've been told, by some one with longer experience with CISCO phones 7960 that if some one try to just replace in the handset the microphone inside with one form a cheaper traditional phone will get this VoIp hardphone working perfect. But in my case i didn't try that. If someone has a SPA942 on their own lab and can try this without damaging the phone would be nice info to share, I believe! Best regards, Marco Mouta On 1/17/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: I too seem to have the same problem, dont know about poor quality but its certainly not loud enough, I have to put my mouth to the microphone, otherwise the other end reports they cannot hear me. This does however seem to do a good job to cancel out the background noise In the SIPura setup change the packet size from .3 to .2. ___ --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 ___ --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] two-level administration tool for Asterisk (reposting)
I like the idea of Virtual PBX, but I don't like python language. Are there other implementations ? I'd like some java or php thing. On 1/16/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Jan 16, 2007 at 10:18:05AM +0500, Kate Kretz wrote: Dear Sirs, let me repost my question again, probably the last one was lost in a huge amount of messages during weekend. I'm actually looking for web-based tool which can do two level of administration: 1) high level, Administrators, can create domains 2) lower level, Users, can manage extensions within certain domain. much like asterisk2billing. so, I want users to manage their things within Asterisk not affecting other users. http://destar.berlios.de/ . What you call domain is called there virtual pbx. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 ___ --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] two-level administration tool for Asterisk (reposting)
Freepbx GUI let's you create different administrators with different permissions! On 1/17/07, Kate Kretz [EMAIL PROTECTED] wrote: I like the idea of Virtual PBX, but I don't like python language. Are there other implementations ? I'd like some java or php thing. On 1/16/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Jan 16, 2007 at 10:18:05AM +0500, Kate Kretz wrote: Dear Sirs, let me repost my question again, probably the last one was lost in a huge amount of messages during weekend. I'm actually looking for web-based tool which can do two level of administration: 1) high level, Administrators, can create domains 2) lower level, Users, can manage extensions within certain domain. much like asterisk2billing. so, I want users to manage their things within Asterisk not affecting other users. http://destar.berlios.de/ . What you call domain is called there virtual pbx. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto: [EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 ___ --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 ___ --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] two-level administration tool for Asterisk (reposting)
On Wed, Jan 17, 2007 at 11:47:35AM +, Marco Mouta wrote: Freepbx GUI let's you create different administrators with different permissions! But can you separate the permissions by context/domain? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] windows mobile 5 softphone for square screen devices
Cant remember the url but I googled it. Xten also without luck.. the main problem is the 240x240 screen... |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of mitcheloc |Sent: Wednesday, January 17, 2007 1:48 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] windows mobile 5 softphone for square screen devices | |I've been trying the SJPhone with no luck. Where did you download the |Xten version from? | |On 1/16/07, Anton Krall [EMAIL PROTECTED] wrote: | Guys, anybody has seen or is using some kind of softphone on any square | screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they | do work on Wm5 but they are designed for standard screens, anybody using | anything on square ones? | | | | ___ | --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 | | | |-- | |Mitchel Constantin |Snap - A desktop user interface for Asterisk |www.snapanumber.com |___ |--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 ___ --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] two-level administration tool for Asterisk (reposting)
My mistake Tzafrir, you are right! On 1/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jan 17, 2007 at 11:47:35AM +, Marco Mouta wrote: Freepbx GUI let's you create different administrators with different permissions! But can you separate the permissions by context/domain? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 ___ --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] windows mobile 5 softphone for square screen devices
Anton Krall a écrit : Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones? We are using PPCIAX. -- Daniel ___ --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] TDM404B VS TDM2401B
Same result, more FXO interfaces. De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Al Envoyé : 17 janvier 2007 00:39 À : asterisk-users@lists.digium.com Objet : [asterisk-users] TDM404B VS TDM2401B Hi List, any good comparison between TDM404B and TDM2401B i'm not very happy with TDM404B voice quality, low volume and sometimes echo. I was wondering if any of you guys have good experience with TDM2401B. thanks! ___ --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] Refreshing DNS lookups
housi mueller wrote: The dnsmgr in Aterisk 1.4.0 seems not to work. I enabled DNS lookups in dnsmgr.conf but after reloading the conf files * never refreshes DNS lookups. Any ideas how to debug this issue? The DNS manager is not used very much in Asterisk 1.4 at all; don't expect it to provide any benefits at this point. ___ --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] Error on answer a SIP 401 message
kjcsb wrote: I had something vaguely similar. Asterisk was replying on the wrong interface/network card. Might be worth checking. Asterisk does not choose (or have any control over) which interface is used for packet transmission. That is the responsibility of your operating system's IP stack. ___ --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] Really Big Queues
Hello Chris, we have a number of clients who deployed very large CCs over the 200 agent range. Your idea #1 is pretty sound and I believe that's what most people are doing. I would like to add a couple of points of attention: - having hundreds of agents on a box means a lot of synchronous audio flowing in and out, so you don't want to save on the ethernet hardware :) - think about a passive SIP monitoring for call recording, so that you can have a different box (or set of boxes) to handle that without slowing down the overtaxed ethernet connections of the queueing servers. I hope this helps, l. In data Wed, 17 Jan 2007 02:32:35 +0100, Christopher Snell [EMAIL PROTECTED] ha scritto: Hi, How do you folks handle really large queues (350+ simultaneous callers) in your Asterisk PBXes? We're going to be bringing in around 16 PRIs' worth of inbound callers, doing skills-based routing, and queuing them up for approximately 200 agents. What's the best way to handle all of these callers? We want to record the calls and we'll probably use the ramdisk method that has been discussed on this list. Here's some ideas that I'm considering: Idea #1: Use servers with (2) Digium 4-port PRI cards, running Asterisk, as media gateways. From here, send calls to 2 or more Asterisk queue servers. For each incoming call, run an AGI on the media gateways that determines which queue server is least loaded. Send this incoming call to the queue server over an IAX2 trunk. The problem with this method is that the queues are not unified; if one queue server suddenly has available agents, queued callers on the other queue server cannot be (easily?) transfered to the server with available agents. Also, running an AGI for each incoming call is lame and slow. Idea #2: Use 3com VCX V7122 media gateways to terminate the PRIs and send the calls to a load balanced pair of SER proxies. These proxies will somehow keep track of the state of the Asterisk queue servers and redirect the incoming calls to the least loaded (most available) queue server. The problem with this method is that, by using SIP, we'll probably see higher interrupt load on the Asterisk queue servers. Additionally, I'm not a SER expert yet and I have no idea how to get SER to monitor the state of the Asterisk queue servers. As with Idea #1, the queues are also not unified, which sucks. Idea #3: ??? (profit!) Do you fine folks have any ideas or suggestions? thanks, Chris -- Home of QueueMetrics - http://queuemetrics.loway.it ___ --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] 2 Questions: Answer with music don't work and Voicemail direct access ?
Hi I have two small question, if you can help me ;=) Problems with Answer+Music my extension: [Cal-In] exten = _81120,1,Goto(C-Internal,100,1) exten = _81121,1,Goto(C-Internal,200,1) [C-Phibee] exten = 100,1,Ringing exten = 100,2,Wait,1 exten = 100,3,Answer exten = 100,4,Dial(SIP/201SIP/200,30) exten = 100,5,Hangup exten = 200,1,Ringing exten = 200,2,Wait,1 exten = 200,3,Answer exten = 200,4,Dial(SIP/200,25,tm) exten = 200,5,Hangup With this extension, when a incoming call are received : If my customer have call 081120, that's answer and Ring If my customer have call 081121, he have a answer, he have a music I don't know why the 081120 don't have the music for wait that i am answer ... Second Question: It's possible to put into the extension, for access to the VoiceMail, the extension of the caller ? exten = 500,1,VoiceMailMain(@Home) Actually, when i call the 500, he want know my mailbox ID and after password ... if i call with the post 200, it's possible to access direclty at the password ? Thanks bye ___ --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 registration
Hi all, some body told me that you can make asterisk to register itself to another asterisk server. i just want to know whether it really can be done or not. i have googled a lot but no answeres. -- Regards Rizwan Hisham Software Engineer ___ --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 registration
You can by using a register entry in either SIP or IAX connections. Try searching voip-info.org for the details of sip.conf and iax.conf and also connectiong 2 asterisk servers. On 1/17/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, some body told me that you can make asterisk to register itself to another asterisk server. i just want to know whether it really can be done or not. i have googled a lot but no answeres. -- Regards Rizwan Hisham Software Engineer ___ --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 -- Bruce Nortex Networks ___ --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] 2 Questions: Answer with music don't work and Voicemail direct access ?
Hi,
Am 17.01.2007 um 15:07 schrieb Noc Phibee:
Problems with Answer+Music
my extension:
[Cal-In]
exten = _81120,1,Goto(C-Internal,100,1)
exten = _81121,1,Goto(C-Internal,200,1)
[C-Phibee]
exten = 100,1,Ringing
exten = 100,2,Wait,1
exten = 100,3,Answer
exten = 100,4,Dial(SIP/201SIP/200,30)
exten = 100,5,Hangup
exten = 200,1,Ringing
exten = 200,2,Wait,1
exten = 200,3,Answer
exten = 200,4,Dial(SIP/200,25,tm)
exten = 200,5,Hangup
With this extension, when a incoming call are received :
If my customer have call 081120, that's answer and Ring
If my customer have call 081121, he have a answer, he have a
music
I don't know why the 081120 don't have the music for wait that
i am answer ...
I guess you simply did a mistake in the Goto. It points to the C-
Internal context but you want to jump to C-Phibee.
It's possible to put into the extension, for access to the
VoiceMail, the extension of the caller ?
exten = 500,1,VoiceMailMain(@Home)
Actually, when i call the 500, he want know my mailbox ID and after
password ...
if i call with the post 200, it's possible to access direclty at
the password ?
Yes:
exten = 500,1,VoiceMailMain(${CALLERID(num)[EMAIL PROTECTED])
But I am not sure if you really want to use @Home here. But that
depends on you voicemail.conf
BTW: With exten = 500,1,VoiceMailMain(${CALLERID(num)[EMAIL PROTECTED],s) you
can even skip the password question.
Stefan
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Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
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[asterisk-users] dtmf problem -- second part
I realize I cannot use inband audio for phones (voicemail and internal ivr, password for external trunks and other thing not working) So I put everywhere rfc2833. Doing this, anyway, make any EXTERNAL IVR NOT working. I see a lot of posts about this, but no solution, becouse using inband audio (which works for outside...) breaks inside IVR Is it possible to define to use inband audio ONLY on the misdn trunks I am using to connect to PSTN ? Moreover, using rfc2833 shouldn't be asterisk to create the dtmf when trunked outside ? Thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --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] php agi - first phrase truncated, all others fine
Greetings,
I have never done any agi programming, but my first thought is maybe you need
a wait statement after answering?
John Beaman
Telecom Specialist
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 1/15/2007 10:53:51 AM
I have the following code. When I call the extension, it either ignores the
first Hello there everyone, or says hello and moves on sometime stoping
before it finishes hello. The rest of the text reads fine. Anyone else
have this issue??
Thanks!
require('/var/lib/asterisk/agi-bin/phpagi.php');
$agi = new AGI();
$agi-answer();
$agi-swift(Hello there everyone );
$agi-swift(Please press 1 for a search .);
$result= $agi-get_data('beep',3, 1);
$zip= $result['result'];
$agi-swift(That concludes your call. Thank you, Good bye .);
$agi-hangup();
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Re: [asterisk-users] Asterisk registration
On 1/18/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, some body told me that you can make asterisk to register itself to another asterisk server. i just want to know whether it really can be done or not. i have googled a lot but no answeres. -- Regards Rizwan Hisham Software Engineer For servers A and B, You need to create a user a/c in say Svr A like rizwan with pwd 1234 and then in svr B sip.conf, put in a line register = rizwan:[EMAIL PROTECTED] You can now create a trunk that uses this a/c to SvrB to terminate calls there See http://www.asterisk.org/doxygen/1.4/Config_sip.html for more info HTH ___ --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] force ulaw passthrough if call from modem extension?
Tried that, it didn't work but maybe I didn't configure it right. Anyways how can I route all outgoing calls from that specific extension to use that trunk? Thanks. On 1/16/07, Tim Panton [EMAIL PROTECTED] wrote: On 16 Jan 2007, at 19:56, Victor Perez wrote: I have Teliax trunk set to ulaw and g729 and I have a modem/fax extension from a sipura forced to ulaw. When the call goes out through Teliax IAX trunk, asterisk transcodes to g729. Is there a way to tell asterisk not to transcode calls from/to a specific extension? try creating a separate (duplicate) entry in iax.conf for the teliax connection disallow 729 on that trunk and use it for fax/alarm calls. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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 ___ --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] Callback/ringback
Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is busy or unavailable then Asterisk should inform the first user that the number they dialed is busy and hangup the call. Once the second caller is available again then Asterisk should initiate a call back to both the users and connect them. Any ideas on how to achieve this will be appreciated. Thanks, Richard . ___ --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] Monitor or log peer performance
In a couple different locations I have some clients that are having intermittent problems. All of my other customers aren't complaining of issues. Whenever I conduct a test, everything is fine. No call quality issues to speak of. What can I do to log\monitor these clients so I can troubleshoot this issue? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --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
