[asterisk-users] Unable to execute 'dahdi_scan /etc/asterisk/dahdi_scan.conf'
Hi Can anyone help me with this error Unable to execute 'dahdi_scan /etc/asterisk/dahdi_scan.conf' i am using asterisk-gui 2.0 , asterisk 1.8, dahdi 2.6 and also my call reached the destination but no voice is coming from destination my voice reflects back thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime peers and trunks coming from the same IP
Hi Sammy go Can you help me with my problem I have asterisk 1.8 and i am using asterisk-gui 2.0, and in asterisk-gui 2.0 the voice prompt menu which is used for custom voice recording for IVR is not working and not recording. Can u tell me how to defualt this feature. thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] upgrading from asterisk 1.4 to 1.6
Hi I have installed asterisk 1.4 and asterisk-gui 2.0, the problem is that it cannot upload the .gsm which i record through voice menu prompt, it gives error uploading is supported in asterisk 1.6 or higher. Can anyone help me? thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)
Hi I am new to asterisk 1.4 can someone tell about how to enable the video conference in asterisk-gui 2.0. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)
Actually i want to know that how i configure the asterisk for video confernce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Double ## Feature to make another call during current call
Dear All Please guide me how can i implement the feature in which when caller and called parties are in conversation and calling party presses double ##, asterisk hangs up called party and take the calling party to some context (or even the current context) in the dialplan where caller could dial another number. Or is there any other way to do this? please guide me!!! Thanks alot, your guidance will be much appreciated. -- Best Regards Atif Razzaq http://atif-razzaq.blogspot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] disconnecting destination channel
Dear All I am using Asterisk 1.4.17 in a calling card application. Following description explains the usage: A call/request hits asterisk from an ip xxx.xxx.xxx.xxx which opens a channel for this ip (Lets call it Channel A). Asterisk answers the call and play IVRs first asking the PIN and then destination number in an AGI making use of radius server for authentication/authorization. Once done with that, Asterisk uses 'Dial' application to move forward sending it to ip yyy.yyy.yyy.yyy which opens another channel to this ip (lets call it channel B). I want to include '##' feature (for making another call), when caller presses ## during a call, destination channel B is hanged up (i need to hangup the destination channel to billing purposes) keeping originating channel A alive and another AGI application is triggered for caller to enter another destination number. Im using feature.conf's application map something like below: followupcall = ##,peer/caller,Hangup At least feature is working but hanging up both channels...Problem!!! If there is any other application to hang up the destination channel, what is that? Also what is the status of originating channel? Where should the call to second AGI be put in the dial plan? I hope you guys understand my problem/issue. Please guide me, thanks alot in advance. -- Best Regards Atif Razzaq http://atif-razzaq.blogspot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NAT
I am interested to configure my linux box for a server for my call center. What sort of NAT/IPTABLES I need to implement on my server? I have just masqurate the box and it is not workingbut i can have local calls .. Thanks in advance. atif ___ --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] compain
I have configured Asterisk and can have calls from one ext to another.But from where i can get the call center compains to make calls for my call center setup? and where/how that sort of stuff will be placed to get the info for the agent? Thanks in advance. atif ___ --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] help about modem
Hi, Sorry if question is stupid..As i m just new to asterisk.. I need help in the following schenerio.. Actually i want to transfer incoming call from PSTN to any PC in the LAN. Can i use modem for this purpose and also need help in configuration for this schenerio.woul any one plz give configuration sample reagarding my problem.. Thanks in advance Best Regards ___ Atif Amin ___ --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-1.2.1.tar on Suse Linux 9
Hi, I am trying to install asterisk on Suse 9. I downloaded asterisk-1.2.1.tar and untar this package. I am following the README and the installation instruction to run make ans make install. But I can not find any make or make install in the directory asterisk-1.2.1. Can any one please help me how can I install asterisk-1.2.1 on Suse? What am I missing? Regards ank ___ --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 with EWSD v16
Dear Gulzar, Thank you for your reply, I am using same configs. I have tried both 0 1 in timing but no luck. I will try again with 'timing' parameter = 1 in zapata.conf best Regards, -- Atif Rasheed Gulzar Hussain wrote: I am using EWSD's PRIs and I am not having this problem my configs are Zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us Zapata.conf [channels] language=en context=ext-acd switchtype=euroisdn signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes group=1 channel = 1-15 channel = 17-31 pridialplan=private prilocaldialplan=private overlapdial=yes usecallerid=yes hidecallerid=no immediate=no usecallingpres=no --- Atif Rasheed [EMAIL PROTECTED] wrote: if any EWSD guru out there..please help ??? Hello all, I am running Asterisk with Digium E1 card with zaptel, libpri, asterisk cvs v1-2. My server is interfaced with EWSD v16 using a PRI on E1. I am running into a problem that at my telco's end alot of trunks are getting BPRM (Block permanant) status. I am not sure why EWSD is blocking trunks. config at my end::: coding = hdb3 format = ccs,crc4 signalling = euroisdn, pri_cpe config at my telco's end coding = hdb3 format = crc4mf signalling = euroisdn, pri_net Is there any EWSD guru around who can explain why trunks are getting BPRM status in EWSD switch. I will really appriciate your help Thank you -- Atif Rasheed ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Asterisk as a gatekeeper
Asterisk can only be configured as a GW AFAIK, what ever flavor of H323 you use with Asterisk it will not work as a GK. Atif rommel malana wrote: Hello, Right now i'm trying to set-up a gatekeeper and i'm having a hardtime doing it, what i'm thinking is instead of having a gatekeeper i'll use the asterisk to be a gatekeeper. Can the asterisk be a gatekeeper? Thanks a lot, Rommel ___ --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 with EWSD v16
Hello all, I am running Asterisk with Digium E1 card with zaptel, libpri, asterisk cvs v1-2. My server is interfaced with EWSD v16 using a PRI on E1. I am running into a problem that at my telco's end alot of trunks are getting BPRM (Block permanant) status. I am not sure why EWSD is blocking trunks. config at my end::: coding = hdb3 format = ccs,crc4 signalling = euroisdn, pri_cpe config at my telco's end coding = hdb3 format = crc4mf signalling = euroisdn, pri_net Is there any EWSD guru around who can explain why trunks are getting BPRM status in EWSD switch. I will really appriciate your help Thank you -- Atif Rasheed ___ --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 with EWSD v16
if any EWSD guru out there..please help ??? Hello all, I am running Asterisk with Digium E1 card with zaptel, libpri, asterisk cvs v1-2. My server is interfaced with EWSD v16 using a PRI on E1. I am running into a problem that at my telco's end alot of trunks are getting BPRM (Block permanant) status. I am not sure why EWSD is blocking trunks. config at my end::: coding = hdb3 format = ccs,crc4 signalling = euroisdn, pri_cpe config at my telco's end coding = hdb3 format = crc4mf signalling = euroisdn, pri_net Is there any EWSD guru around who can explain why trunks are getting BPRM status in EWSD switch. I will really appriciate your help Thank you -- Atif Rasheed ___ --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: binding asterisk-h323 on two interfaces
I have cvs-head of Aug-2. README has no information on how to bind asterisk-h323 on multiple interfaces. actually this was my question that can we bind asterisk-h323 on multiple interfaces ? as h323.conf says that bindaddr should contain a single valid IP. if we bind h323 to 0.0.0.0 as we do in SIP, it sends 0.0.0.0 as it is to the caller and callee. Use cvs -head code from the last day or two and read the README. Jeremy McNamara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can asterisk send Remote-Party-ID header ???
Hello all, Kevin P Fleming once said that a patch will be released very soon to send Remote-Party-ID header from Asterisk. and this was said probably in Feburary. is that patch released yet or not ? if some please comment, I will really appriciate Regards, -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 vs chan_oh323 chan_ooh323
hello there, can somebody please comment which one of these channel drivers will give best output doing g729|g723 pass-thru. only pass-thru is needed no transcoding. please share your experience. if somebody has some figures (simultanous calls using a certain channel driver) it will be apericiated. I have installed chan_h323 (by McNamara) and its working fine with me. I just want to know if I run this driver on a Dual-Xeon machine. can it handle 500 or 500 simultanous calls in pass-thru mode. Regards, -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] looking for some draft (sip - iax2 mapping)
hello all, is there any draft available for sip-iax2 mapping. I mean sip 4XX server failure messages, 5XX Server Failure messages. how these SIP messages are mapped to IAX messages thank you -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] live monitoring of SIP calls chan_spy
hello there, I have searched lists about an application chan_spy, people talked about it on lists that we can use it to monitor sip to sip calls. but I am unable to find any clue of it. can some one please tell me from where I can get this chan-spy application thank you regards, -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk not relaying back the SIP response messages
HI all, I have the following setup running: EP---Calling Asterisk---Relaying Asterisk---Softswitch--- PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling Asterisk. Now the problem is that error responses from the Softswitch to the Relaying Asterisk are not relayed back to the Calling Asterisk. Instead a 403 forbidden error message is sent back to the Calling Asterisk whatever the error response (503, 484, etc). Is there a way to relay back error responses through configuration scripts or do I have to dig in the source code -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC dimensioning
hello there, any one who used ASTCC in a real enviroment, or has successfully handled above 1k simultanous calls. need some evalution of ASTCC. if any one has such an experience please share it with the rest thank you Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * behaviour in agentcallbacklogin
when an agent logs in using AgentCallbackLogin(), during a call when agent presses * call is hanged up. how can I get rid of this behaviour. that nothing should happen by pressing *. thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cvs stable
on the asterisk site, it was stated while ago, how to download stable version. like cvs checkout -r v1-0_stable asterisk-addons zaptel libpri but now it's not their. is stable-version removed from the CVS ? or is their some different procedure ? thank you -- Atif ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pattern matching problems
this is from my extensions.conf, the first three patterns are for
toll-free numbers, and fourth pattern is for other numbers, where an AGI
is called for authentication.
now when I dial 011448000664327 if falls into the fourth pattern, where
as it should be matched by the first pattern. Any suggestions
1 - exten = _01144800XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
2 - exten = _01144808XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
3 - exten = _01144500XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
4 - exten = _011.,1,AGI(iax.agi)
4 - exten = _011.,2,Dial(${MAG}/${EXTEN:3},45,tT)
4 - exten = _011.,103,playback(no-service)
thank you
--
Atif
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[Asterisk-Users] Re: pattern matching problems
thank you people for your help, I have done it, and in a different way,
like
exten = _01144800XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _01144808XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _01144500XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _011X.,1,AGI(iax.agi)
exten = _011X.,2,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _011X.,103,playback(no-service)
I made the _011. more precise, I should say
--
Atif
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[Asterisk-Users] Re: SIP Provider in India/Pakistan/Bengladesh
PTA(Pakistan Telecommunication Authority) has recently issued LDI licences to number of contenders and use VoIP. noone yet has announced but very soon someone from them will announce SIP termination in Pakistan. can't say anything about India/Bengladesh -- Atif ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Asterisk-Users]help me in voice jittering problem
dear all users hi !! , I m using Asterisk as a call manager, i have made two windows clients and a linux server on which asterisk is running , calls are succesfully authenticate in asterisk ... but the problem is with voice jittering when , i record my voice and then play it (by using Playback command)the voice contains so much jittering and cant properly listen what is the message ??? plzzz reply me how to solve this problem i will be very grateful to u1 Atif Azhar
[Asterisk-Users] sip phone, receiving calls but not placing any call
Hello there, I am configuring a sip-phone, it is receiving calls but its not placing calls. sip debug shows that asterisk received digits from phone. but why its not placing calls please help I have dialed 13 from sip-phone, here is some sip-debug INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bKfLZ1GRUt2 Max-Forwards: 70 From: chinee sip:[EMAIL PROTECTED];tag=82veOQ0zKConAx6y To: 13 sip:[EMAIL PROTECTED] Call-ID: y2gsu70CXGySlU0s CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 191 thank you -- Atif ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip phone configuration problem
I am configuring a sip-phone, receing calls, excellent voice quality. but it does not place calls, please, can some one sort out. here is my debug output, and below that is sip-debug, Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0 Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'iiasPlzFribMJMcW' of Response 1: Found Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0 Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'UCWmUU1tF0s6roEx' of Response 2: Found Jul 16 11:34:47 DEBUG[163850]: Auto destroying call 'iiasPlzFribMJMcW' Jul 16 11:34:47 DEBUG[163850]: Auto destroying call 'UCWmUU1tF0s6roEx' **SIP-DEBUG** Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bKAtoCh7hGS Max-Forwards: 70 From: chinee sip:[EMAIL PROTECTED];tag=Zlq179E4Jf8KX2lB To: 13 sip:[EMAIL PROTECTED] Call-ID: 1e020TNnX5IvcvFu CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 221 v=0 o=- 0 0 IN IP4 192.168.0.187 s=- c=IN IP4 192.168.0.187 t=0 0 m=audio 1400 RTP/AVP 0 8 4 18 0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 telephone-event 12 headers, 11 lines Using latest request as basis request Sending to 192.168.0.187 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 0 Peer RTP is at port 192.168.0.187:0 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format telephone-event Capabilities: us - 0x10f(G723|GSM|ULAW|ALAW|G729A), peer - audio=0x109(G723|ALAW|G729A)/video=0x0(EMPTY), combined - 0x109(G723|ALAW|G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bKAtoCh7hGS From: chinee sip:[EMAIL PROTECTED];tag=Zlq179E4Jf8KX2lB To: 13 sip:[EMAIL PROTECTED];tag=as51de164a Call-ID: 1e020TNnX5IvcvFu CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=50b81cdd Content-Length: 0 to 192.168.0.187:5060 Scheduling destruction of call '1e020TNnX5IvcvFu' in 15000 ms Found user 'chinee' Atif Sent via the WebMail system at convergence.com.pk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: How to differentiate a *busy* call from not available?
IsChanAvail() application might help Atif Sent via the WebMail system at convergence.com.pk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: MeetMe Improvement
is there any option of inviting some one to conference, I mean, I press * for menu, then system asks me to invite some one dial 1, and then asks me to dial the extension of that person, and then call is placed to invite that person to conference. Thank you Atif Sent via the WebMail system at convergence.com.pk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems compiling shadydial-asterisk on gentoo
hello there: did some one compiled shadydial with asterisk on gentoo successfully, if some one plz help me I am getting compilation errors during asterisk compilation after replacing the files provided with shadydial thank you here is my log, please help gcc -pipe -I=/usr/local/pgsql/include -pipe -Wall -Wstrict-prototypes -Wmissing -prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GN U_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD- 05/19/04-04:15:26\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIB DIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\ /var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk \ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk /modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_ PRI -DIAX_TRUNKING -DCRYPTO -fPIC -c -o chan_agent.o chan_agent.c chan_agent.c: In function `agent_hangup': chan_agent.c:566: warning: implicit declaration of function `ast_say_digit_str' chan_agent.c: In function `agent_new': chan_agent.c:736: warning: assignment from incompatible pointer type chan_agent.c:754: error: too many arguments to function `ast_queue_frame' chan_agent.c: At top level: chan_agent.c:787: warning: function declaration isn't a prototype make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/usr/src/cvs-src/asterisk/channels' make: *** [subdirs] Error 1 Sent via the WebMail system at convergence.com.pk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAP hangup not working with siemens HICOM
Hello everybody, any configure asterisk with siemens HICOM, or HIPATH. I am facing asevere hangup issue. two servers at station-1 and station-2 both with 8 fxo lines. extensions from siemens are plugged into the fxo cards. now problem is that, if call is made from station-1 to station-2 and then if caller hanged up, 80% chances are that asterisk will not hangup the call, it will remain stucked in the asterisk.\ will playing with the busy tone in indications.conf sort out this problem? Thanks Atif Sent via the WebMail system at convergence.com.pk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: X100P keeping PSTN line Offhook
What did you set your busy count variable to when the calls started to drop? I had the same issue until I changed it to 6 from 4 and so far everything seems to be working fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shahid Sent: Monday, May 10, 2004 7:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: X100P keeping PSTN line Offhook Tom, Rich and Atif, Regarding your responses, 1. I have previously tried the callprogrees=no. Doesnt solve the problem. 2. If busydetect=yes, calls to PSTN get droped in the middle of the conversations. 3. Havent looked into the MOH thingy. This feature has caused me other problems. Thinking of turning it off altogether. Anyone has any ideas about alternatives ? Thanks for all your help guys. Regards -shahid Shahid [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! pbx1*CLI zap show channel 1 Channel: 1 File Descriptor: 31 Span: 1 Extension: Context: bell Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Offhook = zapata.conf == busydetect=no musiconhold=default group=1 pickupgroup=1 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel = 1 pickupgroup=1 immediate=no signalling=fxs_ks callerid=asreceived channel = 2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI.pm wait_for_digit() not working for me!!!
Hello everybody!!!
I really need your help guys, I am using the AGI mode in
meetme application, and I want that AGI should wait for an input from the
client/user i.e. a digit and then proceed, but I have used that AGI function
agi-wait_for_digit(), but no usemy agi just passes, or ignores this
function,
where AGI should stop here and wait for the input
.my extension in my dialplan.
exten = 21,1,answer
exten = 21,2,meetme(21|pb)
..and here is my AGI...
#!/usr/bin/perl -w
#use strict;
$aginame=conf-background.agi;
use File::Copy cp;
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$char=0;
#while(1)
{
#$AGI-exec('WaitExten','25000');
#$char =
$AGI-receive_char('600');
$char=chr($AGI-wait_for_digit('600'));
print STDERR
input form rec char : $char\n;
if($char eq
*)
{
print STDERR Dialing your number\n;
$srcfile=/tmp/mycall;
$dstfile=/var/spool/asterisk/outgoing/mycall;
open(MYCALL,$srcfile) || die Cant't open file :$srcfile
$!\n;
print MYCALL Channel:IAX2/bali:[EMAIL PROTECTED]/[EMAIL PROTECTED];
print MYCALL MaxRetries:2\n;
print MYCALL RetryTime:60\n;
print MYCALL WaitTime:30\n;
print MYCALL Context:atif\n;
print MYCALL Extension:22\n;
print MYCALL Priority:1\n;
close MYCALL;
#
cp($srcfile,$dstfile);
print STDERR dialing complete...\n;
}
RE: [Asterisk-Users] X100P keeping PSTN line Offhook
Try enabling busy detect and set it to a value between 4 and 6. If you set it too low you might start getting random call drops. I think this problem is due to some providers allowing only the called party to hang up. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shahid Sent: Saturday, May 08, 2004 6:20 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X100P keeping PSTN line Offhook Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! pbx1*CLI zap show channel 1 Channel: 1 File Descriptor: 31 Span: 1 Extension: Context: bell Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Offhook = zapata.conf == busydetect=no musiconhold=default group=1 pickupgroup=1 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel = 1 pickupgroup=1 immediate=no signalling=fxs_ks callerid=asreceived channel = 2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] meetme conf-background.agi
Hello there!
Somebody tried the meetme|b which initiates the conf-background
AGI
Actually I want to originate another call from a conferencemy
AGI originates the call and connects it to the conference, but the call is nowhere
My extension
exten = 21,1,meetme(21|pb)
and my AGI
#!/usr/bin/perl -w
$aginame=conf-background.agi;
use File::Copy cp;
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
print STDERR Dialing your number\n;
$srcfile=/tmp/mycall;
$dstfile=/var/spool/asterisk/outgoing/mycall;
open(MYCALL,$srcfile) || die Cant't
open file :$srcfile $!\n;
print MYCALL Channel:Zap/1/13\n;
print MYCALL MaxRetries:2\n;
print MYCALL RetryTime:60\n;
print MYCALL WaitTime:30\n;
print MYCALL Context:default\n;
print MYCALL Extension:22\n;
print MYCALL Priority:1\n;
close MYCALL;
cp($srcfile,$dstfile);
#used to hold the AGI, otherwise it quits
$AGI-get_data('ccs-getnumber','1','2');
print STDERR dialing complete...\n;
Some one can sort out, where things are going wrong
Thank you
Atif
35,1 Top
[Asterisk-Users] loopstart,kewlstart,groundstart
kindly tell me what is difference b/w loopstart, kewlstart, groundstart for FXO or FXS devices Thank you -- Atif Rasheed Convergence (Business Systems) http://www.convergence.com.pk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callgroup pickupgroup and zap problem!!!
I have successfully configured and tested the callgroup and pickupgroup for my ZAP interface.. but suppose I have 12 FXS interfaces, all are in a same pickup and callgroup, and there is only a single access code to pick a call i.e. *8, and if at a time multiple interfaces are ringing i.e. Zap/1,Zap/2,Zap/3 etc, and I dial *8 from my set Zap/5, and I want to pickup call on my choice among those 3or4 calls how could I do that... Thank you -- Atif Rasheed Convergence (Business Systems) http://www.convergence.com.pk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: FreeBSD or Linux
no doubt; 'Gentoo' Linux.Asterisk life partner -- Atif Rasheed Convergence (Business Systems) http://www.convergence.com.pk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to disable zap debug!!!
how to disable this DEBUG information... I am getting this on Asterisk CLI --- Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception: Exception on 22, channel 4 Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:2765 zt_handle_event: Got event On hook(1) on channel 4 (index 0) Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:1133 zt_disable_ec: disabled echo cancellation on channel 4 Mar 5 16:18:17 DEBUG[426001]: channel.c:2275 ast_channel_bridge: Didn't get a frame from channel: Zap/4-1 Mar 5 16:18:17 DEBUG[426001]: channel.c:2343 ast_channel_bridge: Bridge stops bridging channels Zap/3-1 and Zap/4-1 Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:1715 zt_hangup: Hangup: channel: 4 index = 0, normal = 22, callwait = -1, thirdcall = -1 Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:1133 zt_disable_ec: disabled echo cancellation on channel 4 Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:2095 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/4-1 Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:1076 update_conf: Updated conferencing on 4, with 0 conference users -- Hungup 'Zap/4-1' --- -- Atif Rasheed Convergence (Business Systems) http://www.convergence.com.pk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail not working with mysql !!!
I am a newbie to asterisk if u please sort this out... and kindly tell me how to mail to ur mailing lists...whose archives are on www.mark.net I need some tips on configuration of voicemail with mysql... here is my voicemail.conf **voicemail.conf*** [general] dbhost=localhost dbname=asteriskvmusers dbuser=root format=wav serveremail=asterisk attach=yes maxmessage=60 maxgreet=60 maxlogins=3 [default] 1234 = 7654,Atif Rasheed,[EMAIL PROTECTED] **voicemail.conf*** I have created the databaseasteriskvmusers in mysql and then created the table 'users' in that database. mysql select * from users; +-+-+--+--++---+-++ | context | mailbox | password | fullname | email | pager | options | stamp | +-+-+--+--++---+-++ | default | 1234| 7654 | Atif Rasheed | [EMAIL PROTECTED] | | | 00 | +-+-+--+--++---+-++ but it's not working...i mean when I change the passward through the zap interface it is changed in the file 'voicemail.conf' but database is not effected at all... one more thing which one is newer version, and has mysql support... voicemail or voicemail2 please figure this out... Thank you -- Atif Rasheed Convergence (Business Systems) http://www.convergence.com.pk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Dev digest, Vol 1 #507 - 8 msgs
I need some tips on configuration of voicemail with mysql... here is my voicemail.conf **voicemail.conf*** [general] dbhost=localhost dbname=asteriskvmusers dbuser=root format=wav serveremail=asterisk attach=yes maxmessage=60 maxgreet=60 maxlogins=3 [default] 1234 = 7654,Atif Rasheed,[EMAIL PROTECTED] **voicemail.conf*** I have created the databaseasteriskvmusers in mysql and then created the table 'users' in that database. mysql select * from users; +-+-+--+--++---+-++ | context | mailbox | password | fullname | email | pager | options | stamp | +-+-+--+--++---+-++ | default | 1234| 7654 | Atif Rasheed | [EMAIL PROTECTED] | | | 00 | +-+-+--+--++---+-++ but it's not working...i mean when I change the passward through the zap interface it is changed in the file 'voicemail.conf' but database is not effected at all... one more thing which one is newer version, and has mysql support... voicemail or voicemail2 can someone figure out this... Thank you have you enabled USE_MYSQL_VM_INTERFACE=1 in the asterisk/apps/Makefile ? matteo now I have enabled it and recompiled the asterisk...but still not working can someone figure it out -- Atif Rasheed Convergence (Buisness solutions) http://www.convergence.com.pk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail not working with mysql!!!!
I need some tips on configuration of voicemail with mysql... here is my voicemail.conf **voicemail.conf*** [general] dbhost=localhost dbname=asteriskvmusers dbuser=root format=wav serveremail=asterisk attach=yes maxmessage=60 maxgreet=60 maxlogins=3 [default] 1234 = 7654,Atif Rasheed,[EMAIL PROTECTED] **voicemail.conf*** I have created the databaseasteriskvmusers in mysql and then created the table 'users' in that database. mysql select * from users; +-+-+--+--++---+-++ | context | mailbox | password | fullname | email | pager | options | stamp | +-+-+--+--++---+-++ | default | 1234| 7654 | Atif Rasheed | [EMAIL PROTECTED] | | | 00 | +-+-+--+--++---+-++ but it's not working...i mean when I change the passward through the zap interface it is changed in the file 'voicemail.conf' but database is not effected at all... one more thing which one is newer versionand has mysql support voicemail or voicemail2 can someone figure out this... Thank you -- Atif Rasheed Convergence (Buisness solutions) http://www.convergence.com.pk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cannot configure voicemail with mysql
I need some tips on configuration of voicemail with mysql... here is my voicemail.conf **voicemail.conf*** [general] dbhost=localhost dbname=asteriskvmusers dbuser=root format=wav serveremail=asterisk attach=yes maxmessage=60 maxgreet=60 maxlogins=3 [default] 1234 = 7654,Atif Rasheed,[EMAIL PROTECTED] **voicemail.conf*** I have created the databaseasteriskvmusers in mysql and then created the table 'users' in that database. mysql select * from users; +-+-+--+--++---+-++ | context | mailbox | password | fullname | email | pager | options | stamp | +-+-+--+--++---+-++ | default | 1234| 7654 | Atif Rasheed | [EMAIL PROTECTED] | | | 00 | +-+-+--+--++---+-++ but it's not working...i mean when I change the passward through the zap interface it is changed in the file 'voicemail.conf' but database is not effected at all... one more thing which one is newer version, and has mysql support... voicemail or voicemail2 can someone figure out this... Thank you -- Atif Rasheed Convergence (Buisness solutions) http://www.convergence.com.pk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail not working with mysql!!!!
I need some tips on configuration of voicemail with mysql... here is my voicemail.conf **voicemail.conf*** [general] dbhost=localhost dbname=asteriskvmusers dbuser=root format=wav serveremail=asterisk attach=yes maxmessage=60 maxgreet=60 maxlogins=3 [default] 1234 = 6543,Atif Rasheed,[EMAIL PROTECTED] **voicemail.conf*** I have created the databaseasteriskvmusers in mysql and then created the table 'users' in that database. mysql select * from users; +-+-+--+--++---+-++ | context | mailbox | password | fullname | email | pager | options | stamp | +-+-+--+--++---+-++ | default | 1234| 7654 | Atif Rasheed | [EMAIL PROTECTED] | | | 00 | +-+-+--+--++---+-++ but it's not working...i mean when I change the passward through the zap interface it is changed in the file 'voicemail.conf' but database is not effected at all... one more thing which one is newer version voicemail or voicemail2 voicemailmain or voicemailmain2 and which one is compatible with mysql can someone figure it out... Thank you ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
