[asterisk-users] asterisk speech to text and text to speech?
Dear All Can you please let me know if the asterisk has speech to text and text to speech facilities? Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk dialup connection?
Dear All I need to offer dialup connection for my subscribers. When I put the codec on G.711 the dialup connection will be successful but for the G.723 G.729 it is not. Can you please let me know what are stuffs do I need to have dialup connection when choosing G.723 G.729 codecs? Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2 ?
On Wed, Feb 3, 2010 at 12:17 AM, Ben Dinnerville b...@voicelogic.com.auwrote: This is usually due to an error with the SIP stack not being loaded due to an error - make sure that full logging is on and check your log file and search for ERROR and see if there is any mention to SIP (chan_sip.o etc), alternatively, start asterisk from the command like with asterisk -vdc and watch the output to screen for any errors at startup. Fix the error and SIP will start up. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you very much for your reply . I found my mistake . It was coming from my attempt to copy the old sip.conf extensions.conf onto the new build ones . It seems that it is not possible this way . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2 ?
Dear All On my CentOS 5 server , I have upgraded my Asterisk from 1.4 to 1.6.2 but its CLI help does not show sip and when dialing outward sip it complains as 'sip not implemented' . Can you please let me know what is wrong my case here ? Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2 ?
On Tue, Feb 2, 2010 at 12:40 PM, hadi motamedi motamed...@gmail.com wrote: Dear All On my CentOS 5 server , I have upgraded my Asterisk from 1.4 to 1.6.2 but its CLI help does not show sip and when dialing outward sip it complains as 'sip not implemented' . Can you please let me know what is wrong my case here ? Thank you Sorry . Forgot to mention that I have made use of the following packages for the upgrade procedure : asterisk-1.6.2.1.tar.gz dahdi-linux-complete-2.2.1+2.2.1.tar.gz libpri-1.4.10.2.tar.gz Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson o...@edvina.net wrote: 13 jan 2010 kl. 06.56 skrev hadi motamedi: Dear All I have Asterisk 1.4 installed on my Debian server . I am considering upgrading my Asterisk to the latest version (1.6) . Can you please let me know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk 1.6 ? Please observe that there is no 1.6 version. Previous to 1.6.0, there was 1.0 and 1.2 and 1.4. Then the release policy changed and we had 1.6.0 and 1.6.1 and now (latest) 1.6.2. These three 1.6.x are all very, very different. Also note that none of these are LTS releases, something that was recently introduced. LTS is Long Term Support. 1.4 is classified as LTS, and the next LTS will be version 1.8 - yes, we're correcting the mistake and going back to the old way of numbering releases. Personally, I see the 1.6.x releases as very experimental and don't recommend them for production use. In regards to changes, there has been a massive amount of changes, especially work done by the Digium dev team to rebuild the internal structure of Asterisk to support massive scalability and improve stability of Asterisk. The major new feature is of course faxing, that was introduced in 1.6.0 and has been improved in every release. Please download the new version and read the documentation that covers the CHANGES as well as instructions for upgrading your product. As always, there's no reason to upgrade if there are no features you need. 1.4 is still a supported release. Best regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . I downloaded the Asterisk-1.6.2.0 and left my libpri-1.4 and zaptel-1.4 as they are . After the installation , according to you , I just have the fax feature that is being added . Can you please confirm if nothing wrong in my case? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
On Wed, Jan 13, 2010 at 8:27 AM, Olle E. Johansson o...@edvina.net wrote: My apologies for the multiple copies. Had issues with a mailserver that somehow wasn't talking to DNS properly. Now fixed. It behaved like Asterisk does sometimes, very poor when it can't connect to DNS. Had power outage yesterday and I think that started it all... Meanwhile, I tried to retransmit to find the issue, as I noticed that my mails did not reach the list. Guess what, they all did in the end... ;-) /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you . Receiving your multiple replies is no problem at all . Looking forward your reply on upgrading to Asterisk-1.6.2.0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
On Wed, Jan 13, 2010 at 8:49 AM, Olle E. Johansson o...@edvina.net wrote: 13 jan 2010 kl. 09.26 skrev hadi motamedi: On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson o...@edvina.net wrote: 13 jan 2010 kl. 06.56 skrev hadi motamedi: Dear All I have Asterisk 1.4 installed on my Debian server . I am considering upgrading my Asterisk to the latest version (1.6) . Can you please let me know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk 1.6 ? Please observe that there is no 1.6 version. Previous to 1.6.0, there was 1.0 and 1.2 and 1.4. Then the release policy changed and we had 1.6.0 and 1.6.1 and now (latest) 1.6.2. These three 1.6.x are all very, very different. Also note that none of these are LTS releases, something that was recently introduced. LTS is Long Term Support. 1.4 is classified as LTS, and the next LTS will be version 1.8 - yes, we're correcting the mistake and going back to the old way of numbering releases. Personally, I see the 1.6.x releases as very experimental and don't recommend them for production use. In regards to changes, there has been a massive amount of changes, especially work done by the Digium dev team to rebuild the internal structure of Asterisk to support massive scalability and improve stability of Asterisk. The major new feature is of course faxing, that was introduced in 1.6.0 and has been improved in every release. Please download the new version and read the documentation that covers the CHANGES as well as instructions for upgrading your product. As always, there's no reason to upgrade if there are no features you need. 1.4 is still a supported release. Best regards, /Olle Thank you for your reply . I downloaded the Asterisk-1.6.2.0 and left my libpri-1.4 and zaptel-1.4 as they are . After the installation , according to you , I just have the fax feature that is being added . Can you please confirm if nothing wrong in my case? I'm sorry, I don't understand you. Please check the documentation to find all new features added, the CHANGES file is a good start. If you update to latest Asterisk, I think you should update libpri and change zaptel to Dahdi to get access to the latest features of all packages. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you very much for your reply . I am using from Asterisk:The future of telephony book to install Asterisk . For the new version , according to you , I am using from Asterisk-1.6.2.0 , Libpri 1.4.10.2 , and Dahdi 2.2.0.2 . But for Dahdi installation , the mentioned book does not have any section . Can you please confirm if the Dahdi installation is the same as Libpri or not? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
Dear All I have Asterisk 1.4 installed on my Debian server . I am considering upgrading my Asterisk to the latest version (1.6) . Can you please let me know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk 1.6 ? Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] You won't help me anymore?
Dear All You are not willing to help me anymore ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] You won't help me anymore?
On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra csi...@gmail.com wrote: Sunday, January 10, 2010, 11:24:22 AM, hadi wrote: You are not willing to help me anymore ? Why do you think this? -- Best regards, Gergomailto:csi...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . I am facing with callerId problem on my sip inbound calls , so I strongly need your technical help . Can you please help me ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] You won't help me anymore?
On Sun, Jan 10, 2010 at 2:43 PM, Geoff Lane ge...@gjctech.co.uk wrote: On Sunday, January 10, 2010, Francesco Peeters wrote: Yes, post your question clear and consicely, include all relevant information and snip all unneccessary history. Note that: no reply != not wanting to help... It *is* obviously possible people just do not KNOW the answer!... (Oh what shock and horror!!!) FWIW, he did post his question yesterday. I've just taken a look and one potential issue I've spotted is that the external server he mentions is 192.168.0.139, which is part of the 192.168.0.0/16 netblock reserved for private networks. So while the server might be 192.168.0.139 on it's own LAN, I suspect that won't be its public IP address. Other than that, I suspect there might be an issue with the dialplan. The OP posted an excerpt from his sip.conf but I suspect we'd need his extensions.conf or extensions.ael (whichever or both he's using) before being able to help further. HTH, -- Geoff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you very much for your reply . My Asterisk CallerId issue is as the followings : My Asterisk has sip connection with external sip server and sip inbound and outbound calls are ok . But for the sip inbound calls when the external sip server sends SIP INVITE with CallerId field in the range of my Asterisk sip phones the call will be rejected . For example , please imagine that my Asterisk sip phones are at 667 range so when the external sip server places sip inbound call with SIP INVITE CallerId as say 667 2020 the call will be rejected . But if he modifies his CallerId to say 021 667 2020 (i.e. with area code included) the call will get through . Can you please let me know what is the problem here ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] You won't help me anymore?
On Mon, Jan 11, 2010 at 6:23 AM, Zhang Shukun bit...@gmail.com wrote: you'd better paste your dialplan snip here, in order to get specific help. 2010/1/11 Darrick Hartman dhart...@djhsolutions.com: On 01/10/2010 11:38 PM, hadi motamedi wrote: FWIW, he did post his question yesterday. I've just taken a look and one potential issue I've spotted is that the external server he mentions is 192.168.0.139, which is part of the 192.168.0.0/16 http://192.168.0.0/16 netblock reserved for private networks. So while the server might be 192.168.0.139 on it's own LAN, I suspect that won't be its public IP address. Other than that, I suspect there might be an issue with the dialplan. The OP posted an excerpt from his sip.conf but I suspect we'd need his extensions.conf or extensions.ael (whichever or both he's using) before being able to help further. Thank you very much for your reply . My Asterisk CallerId issue is as the followings : My Asterisk has sip connection with external sip server and sip inbound and outbound calls are ok . But for the sip inbound calls when the external sip server sends SIP INVITE with CallerId field in the range of my Asterisk sip phones the call will be rejected . For example , please imagine that my Asterisk sip phones are at 667 range so when the external sip server places sip inbound call with SIP INVITE CallerId as say 667 2020 the call will be rejected . But if he modifies his CallerId to say 021 667 2020 (i.e. with area code included) the call will get through . Can you please let me know what is the problem here ? It sounds like a dialplan issue where you don't have a pattern which matches 6662020 while you do have something that matches 0216672020. Without seeing the dialplan, we can only guess. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please find attached my dial plan . extensions.conf Description: Binary data -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk CallerId problem?
Dear All My Asterisk has sip connection with an external sip server @192.168.0.139. I have sip inbound and outbound calls as ok . But there is a problem on sip incoming calls . To illustrate the problem , please suppose the sip phone on external sip server dials my Asterisk sip phone @6672019 . Please find below this sip phone profile inside my sip.conf : [6672019] type=friend ;type=peer ;type=user context=sip-outgoing UserName=6672019 secret=uT0Pc4rU callerid=66720196672019 mailbox=6672...@default canreinvite=no host=dynamic nat=no When the sip phone on external sip server calls my sip @6672019 the sip incoming calls can gets through if and only if the external sip server sends the CallerId field in his SIP INVITE everything other than 667 . For example , sending it like 021 667 (with area code included) can gets through . Can you please let me know how can I overcome this as I need to show the correct CallerId on my sip phone ? Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:How to define incoming route for sip?
Dear All
Can you please let me know how can I define incoming route to accept
incoming calls from an external sip server?
I have defined the following profile for my sip phone :
Under sip.conf :
-
[osaka]
type=friend
context=sip-outgoing
host=192.168.0.139
disallow=all
allow=alaw
[6672019]
type=friend
context=sip-outgoing
canreinvite=no
host=dynamic
nat=no
Under extensions.conf :
[sip-outgoing]
include=sip_outgoing
[sip_outgoing]
exten = _XXX,1,Dial(SIP/osaka/${EXTEN})
[line-incoming]
exten = _6XX,1,Dial(SIP/${EXTEN})
Please be informed that the sip outbound toward the external sip server is
quite ok , but sip incoming is not working . Can you please let me know why
my incoming route is not working properly ?
Thank you
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Re: [asterisk-users] Inquiry:How to define incoming route for sip?
On Wed, Jan 6, 2010 at 11:55 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
Hi,
I noticed you always prefix 'Inquiry:' to your questions on the list.
This is implied from the subject line itself, and wastes some space in
the subject line, so I guess it is kind of pointless.
Now to the question itself,
On Wed, Jan 06, 2010 at 10:44:31AM +, hadi motamedi wrote:
Can you please let me know how can I define incoming route to accept
incoming calls from an external sip server?
Just send them there?
I have defined the following profile for my sip phone :
Under sip.conf :
-
[osaka]
type=friend
context=sip-outgoing
host=192.168.0.139
disallow=all
allow=alaw
This looks like a local phone, and you direct all the calls coming from
it to the context 'sip-outgoing' .
[6672019]
type=friend
context=sip-outgoing
canreinvite=no
host=dynamic
nat=no
Likewise this one (though it registers).
Under extensions.conf :
[sip-outgoing]
include=sip_outgoing
[sip_outgoing]
exten = _XXX,1,Dial(SIP/osaka/${EXTEN})
[line-incoming]
exten = _6XX,1,Dial(SIP/${EXTEN})
Could you explain what you actually want to do? Where do you expect
those SIP calls will come from?
Please be informed that the sip outbound toward the external sip server
is
quite ok , but sip incoming is not working . Can you please let me know
why
my incoming route is not working properly ?
I would actually go the other way around. Please try to convince us
(which also implies: convince yourself) that your setup should work.
Please try to explain why an incoming call should work according to your
configuration.
--
Tzafrir Cohen
icq#16849755
jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Thank you for your reply . I want to correctly route the incoming calls
coming from external sip (named in my profile as osaka) to the destination
(that is my Asterisk subscriber sip phone) . To this end , I defined the
osaka profile in my sip.conf and my Asterisk subscriber phone is at 6672019
that I have defined his profile in my sip.conf as well (as you saw it) .
Then , I tried to define the [sip-outgoing] route in my extensions.conf for
rourting my Asterisk sip subscriber outgoing calls toward the external sip
server (named osaka) and it works here . But my [line-incoming] route for
accepting incoming sip calls from external sip server (osaka) toward my
Asterisk subscriber sip phone at 6672019 fails . Can you please let me know
what is wrong in my configuration ?
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Re: [asterisk-users] Inquiry:Asterisk different codec schemes?
On Tue, Jan 5, 2010 at 8:59 AM, Olle E. Johansson o...@edvina.net wrote: 4 jan 2010 kl. 14.46 skrev Kevin P. Fleming: hadi motamedi wrote: Sorry . I didn't get the point clearly . In the SIP Invite message , it says my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C. The remote endpoint responds with a 200 OK, saying my audio stream is at IP y.y.y.y port y, and I choose codec B. Can you please do me favor and let me know if my understanding is right or not ? Thank you No, you are not understanding the SDP offer/answer model properly. If one endpoint offers codecs A, B and C in its SDP, it is willing to *receive* media in those formats. The receiver of that offer can choose to send media to the offerer in any of those formats, at any time. If the answering endpoint includes only codec B in its SDP, then it is willing to *receive* only codec B. In that scenario, it is possible for media to flow from endpoint 1 to endpoint 2 using codec B, and from endpoint 2 to endpoint 1 using codec A (or C), but this will not happen if Asterisk is an endpoint in this scenario. When Asterisk receives a media frame, if the format of that frame is not the format that it is currently sending to the other endpoint, it will switch to that format automatically. If it cannot do so because the other endpoint did not offer to receive that format, then the call's audio will probably fail. This is the reason why I responded before that Asterisk does not support asymmetric formats in a media session. In reality, it is extremely uncommon for a SIP endpoint to want to send media in a format that it is not also willing to receive; in fact, I can't say I've ever seen this situation arise in any testing I've done or in any issues reported in our issue tracker. But it's fairly common to have asymmetric media in the call. If the caller offers A, B and C and the callee responds with B, the caller sends B but the callee might send A. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry . You mean we can have asymmetric codecs in Asterisk ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk different codec schemes?
On Mon, Jan 4, 2010 at 1:46 PM, Kevin P. Fleming kpflem...@digium.comwrote: hadi motamedi wrote: Sorry . I didn't get the point clearly . In the SIP Invite message , it says my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C. The remote endpoint responds with a 200 OK, saying my audio stream is at IP y.y.y.y port y, and I choose codec B. Can you please do me favor and let me know if my understanding is right or not ? Thank you No, you are not understanding the SDP offer/answer model properly. If one endpoint offers codecs A, B and C in its SDP, it is willing to *receive* media in those formats. The receiver of that offer can choose to send media to the offerer in any of those formats, at any time. If the answering endpoint includes only codec B in its SDP, then it is willing to *receive* only codec B. In that scenario, it is possible for media to flow from endpoint 1 to endpoint 2 using codec B, and from endpoint 2 to endpoint 1 using codec A (or C), but this will not happen if Asterisk is an endpoint in this scenario. When Asterisk receives a media frame, if the format of that frame is not the format that it is currently sending to the other endpoint, it will switch to that format automatically. If it cannot do so because the other endpoint did not offer to receive that format, then the call's audio will probably fail. This is the reason why I responded before that Asterisk does not support asymmetric formats in a media session. In reality, it is extremely uncommon for a SIP endpoint to want to send media in a format that it is not also willing to receive; in fact, I can't say I've ever seen this situation arise in any testing I've done or in any issues reported in our issue tracker. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you very much for correcting me . ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk sending dialed digits in one-by-one digit format?
Dear All Further to my previous inquiry regarding Asterisk sending dialed digits in one-by-one digit format when we had ISDN PRI link with the PSTN switch , you told me that we are expected to enable overlap dialing . At now , we have the same configuration but sip connection to the external sip server . Please be informed that the sip inbound outbound is working correctly but we are expected to send the dialed digits in one-by-one digit format . Can you please let me know what is applicable here in our case ? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:How to join Asterisk real time chat?
Dear All Can you please give me guidelines and the link to join Asterisk real time chat to have your online technical support? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk different codec schemes?
On Thu, Dec 31, 2009 at 12:10 PM, Kevin P. Fleming kpflem...@digium.comwrote: hadi motamedi wrote: Can you please let me know if we can have different codec schemes for audio codec in audio codec out ? I mean , in one application , we can have our audio codec input set to G.711 a-law and our audio codec output set to G.711 u-law . I am facing with an application that calls for such a settings . Asterisk does not support asymmetric codec configurations. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry . I didn't get the point clearly . In the SIP Invite message , it says my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C. The remote endpoint responds with a 200 OK, saying my audio stream is at IP y.y.y.y port y, and I choose codec B. Can you please do me favor and let me know if my understanding is right or not ? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk sip ?
On Thu, Dec 31, 2009 at 6:40 AM, hadi motamedi motamed...@gmail.com wrote: Dear All Please be informed that my Asterisk has sip connection to an external sip server but the sip outgoing call will be disconnected for some unknown reasons . Please find attached the debug log . Can you please do me favor and let me know what is the problem that causes the call to immediately being dropped when the called party goes offhook ? Thank you Dear All Please be informed that the problem came from canreinvite=yes settings . It changed to canreinvite=no and the problem solved out. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk festival?
Dear All I want to enable festival text-to-speech . To this end , I added the required lines to festival.scm but when I want to start festival server I face with the following error : #festival --server SIOD ERROR: end of file inside list Closing a file left open: /usr/share/festival/festival.scm Closing a file left open: /usr/share/festival/init.scm festival: fatal error exiting. Can you please let me know what is its meaning ? Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk Dictate?
Dear All Can you please give me more hint on how Asterisk Dictate() works? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk different codec schemes?
Dear All Can you please let me know if we can have different codec schemes for audio codec in audio codec out ? I mean , in one application , we can have our audio codec input set to G.711 a-law and our audio codec output set to G.711 u-law . I am facing with an application that calls for such a settings . Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
On Wed, Sep 9, 2009 at 4:02 AM, Jeff LaCoursiere j...@jeff.net wrote: On Wed, 9 Sep 2009, hadi motamedi wrote: Thank you for your message . But I tried to find it on my server , as the followings : #find / -name sip.cfg -print But it didn't return any result . Can you please let me know where can I find it ? You probably have not setup central provisioning for your Polycom phones. I am guessing you are configuring them from their (horribly crappy) web interface. Although this kind of works, you will not be able to unleash the true power of your phones without setting up central provisioning. Worse you may be running an old version of the firmware, which may have problems. This involves getting the firmware and XML templates from Polycom, which will include the file sip.cfg. You will have to unpack these files on a TFTP or HTTP server, create XML files for each phone, and point the phone to the server to pick it up. There are numerous howtos on the web to set this up. Time for Google! j On Tue, Sep 8, 2009 at 2:58 PM, Steve Edwards asterisk@sedwards.com wrote: On Tue, 8 Sep 2009, hadi motamedi wrote: I sent you a message regarding my problem with Asterisk Call Parking feature and you told me that needs to check the polycom sip.cfg file . But my Asterisk doesn't have sip.cfg file . Can you please let me know how can I overcome ? sip.cfg is not an Asterisk file. sip.cfg should be in the directory the phone downloads it's configuration from. Typically, /tftpboot/ on a tftp server. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dear All Further to this issue that I asked you before , please be informed that I setup for sip calls from my Asterisk console to SJPhone as sip client on an MS Windows machine . All of the configurations are working properly , I mean sip outgoing and sip incoming and voicemail but the call parking . Can you please let me know why I cannot still solve this issue ? It is appearing to me that the Polycom cfg is no longer involved here . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
On Tue, Dec 22, 2009 at 11:41 AM, Dan Journo
d...@keshercommunications.comwrote:
I recommend you follow the detailed install guide in this book and
install all the required support programs etc.
http://downloads.oreilly.com/books/9780596510480.pdf
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*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *hadi motamedi
*Sent:* 22 December 2009 10:47
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS
5.2?
On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby wcse...@selbytech.com
wrote:
And what is the output of the ./configure? Does it generate any
errors?
Thanks,
--Warren Selby
On Dec 22, 2009, at 1:09 AM, hadi motamedi motamed...@gmail.com
wrote:
On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby wcse...@selbytech.com
wrote:
On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi
motamed...@gmail.comwrote:
Please find below the error message that I got when issuing make
install :
[r...@mss-0 asterisk-1.4.26]# make install
make: -F.: Command not found
The configure script must be executed before running 'make'.
Please run ./configure.
make: *** [makeopts] Error 1
And did you run ./configure like the error message says?
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Yes , I did .
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Please find below the output of ./configure :
[r...@mss-0 asterisk-1.4.26]# ./configure
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking how to run the C preprocessor... gcc -E
checking for grep that handles long lines and -e... /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking minix/config.h usability... no
checking minix/config.h presence... no
checking for minix/config.h... no
checking whether it is safe to define __EXTENSIONS__... yes
checking for uname... /bin/uname
checking for gcc... (cached) gcc
checking whether we are using the GNU C
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @192.168.0.139. To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham dcunning...@voisonics.com wrote: Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using sip set debug on might help you with the Host '192.168.0.139' does not implement 'REGISTER' problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi motamed...@gmail.comwrote: On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.comwrote: On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @ 192.168.0.139 . To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users With many thanks , please let me to ask you if I rely upon on my Asterisk 1.4 installation on my CentOS 5.0 and want to have this external sip client on my CentOS server as well so what will be the solution ? The one you told me was for the Laboratory test when the Asterisk on CentOS calls sip client on MS Windows but what will be the solution if the Asterisk on CentOS calls sip client on the same CentOS ? Is there a Voip application on the CentOS that can resemble this external sip client ? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham dcunning...@voisonics.com wrote: Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using sip set debug on might help you with the Host '192.168.0.139' does not implement 'REGISTER' problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi motamed...@gmail.comwrote: On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.comwrote: On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @ 192.168.0.139 . To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I downloaded installed the AsteriskWin32 PBX but it doesn't have sip server functionality . Can you please propose for an alternative to be used on the MS Windows client as external sip server for my Asterisk on CentOS ? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Wed, Dec 23, 2009 at 1:55 PM, David Cunningham dcunning...@voisonics.com wrote: AsteriskWin32 does have SIP server functionality, same as the linux version. I can't think of any reason why having your CentOS Asterisk be both client and server and register with itself wouldn't work. Although I am wondering how much help all this will be in debugging a connection problem to another SIP provider... On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi motamed...@gmail.comwrote: On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham dcunning...@voisonics.com wrote: Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using sip set debug on might help you with the Host '192.168.0.139' does not implement 'REGISTER' problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi motamed...@gmail.comwrote: On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.comwrote: On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @ 192.168.0.139 . To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I downloaded installed the AsteriskWin32 PBX but it doesn't have sip server functionality . Can you please propose for an alternative to be used on the MS Windows client as external sip server for my Asterisk on CentOS ? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit
Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby wcse...@selbytech.com
wrote:
And what is the output of the ./configure? Does it generate any
errors?
Thanks,
--Warren Selby
On Dec 22, 2009, at 1:09 AM, hadi motamedi motamed...@gmail.com
wrote:
On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby wcse...@selbytech.com
wrote:
On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi
motamed...@gmail.comwrote:
Please find below the error message that I got when issuing make
install :
[r...@mss-0 asterisk-1.4.26]# make install
make: -F.: Command not found
The configure script must be executed before running 'make'.
Please run ./configure.
make: *** [makeopts] Error 1
And did you run ./configure like the error message says?
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Yes , I did .
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Please find below the output of ./configure :
[r...@mss-0 asterisk-1.4.26]# ./configure
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking how to run the C preprocessor... gcc -E
checking for grep that handles long lines and -e... /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking minix/config.h usability... no
checking minix/config.h presence... no
checking for minix/config.h... no
checking whether it is safe to define __EXTENSIONS__... yes
checking for uname... /bin/uname
checking for gcc... (cached) gcc
checking whether we are using the GNU C compiler... (cached) yes
checking whether gcc accepts -g... (cached) yes
checking for gcc option to accept ISO C89... (cached) none needed
checking for g++... no
checking for c++... no
checking for gpp... no
checking for aCC... no
checking for CC... no
checking for cxx... no
checking for cc++... no
checking for cl.exe... no
checking for FCC... no
checking for KCC... no
checking for RCC... no
checking for xlC_r... no
checking for xlC... no
checking whether we are using the GNU C++ compiler... no
checking whether g++ accepts -g... no
checking how to run the C preprocessor... gcc -E
checking how to run the C++ preprocessor... /lib/cpp
configure: error: in `/usr/local/asterisk-1.4.26':
configure: error: C++ preprocessor /lib/cpp fails sanity check
See `config.log' for more details.
Do you have gcc and company installed? gxx, g++?
--
Tzafrir Cohen
icq#16849755
jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Yes , I have g++ installed .
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[asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
Dear All I have tried to install the asterisk-1.4 , libpri-1.4 , and zaptel-1.4 on my CentOS 5.2 server , but my installation unsuccessful . When I check for the presence of installed packages , like the followings , I see the output for libpri and zaptel but nothing is seen for asterisk : #whereis asterisk #whereis libpri #whereis zaptel To install asterisk , I tried like the followings : #make clean #./configure #make menuselect #make #make install #make samples Can you please let me know what is wrong in my installation ? Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
On Mon, Dec 21, 2009 at 12:51 PM, Dan Journo d...@keshercommunications.comwrote: Do you have any error logs? What output do you get when you try “make install” with the asterisk package? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please find below the error message that I got when issuing make install : [r...@mss-0 asterisk-1.4.26]# make install make: -F.: Command not found The configure script must be executed before running 'make'. Please run ./configure. make: *** [makeopts] Error 1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby wcse...@selbytech.com wrote:
On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi motamed...@gmail.comwrote:
Please find below the error message that I got when issuing make install
:
[r...@mss-0 asterisk-1.4.26]# make install
make: -F.: Command not found
The configure script must be executed before running 'make'.
Please run ./configure.
make: *** [makeopts] Error 1
And did you run ./configure like the error message says?
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Yes , I did .
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Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby wcse...@selbytech.com wrote:
And what is the output of the ./configure? Does it generate any errors?
Thanks,
--Warren Selby
On Dec 22, 2009, at 1:09 AM, hadi motamedi motamed...@gmail.com wrote:
On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby wcse...@selbytech.comwrote:
On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi motamed...@gmail.comwrote:
Please find below the error message that I got when issuing make
install :
[r...@mss-0 asterisk-1.4.26]# make install
make: -F.: Command not found
The configure script must be executed before running 'make'.
Please run ./configure.
make: *** [makeopts] Error 1
And did you run ./configure like the error message says?
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Yes , I did .
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Please find below the output of ./configure :
[r...@mss-0 asterisk-1.4.26]# ./configure
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking how to run the C preprocessor... gcc -E
checking for grep that handles long lines and -e... /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking minix/config.h usability... no
checking minix/config.h presence... no
checking for minix/config.h... no
checking whether it is safe to define __EXTENSIONS__... yes
checking for uname... /bin/uname
checking for gcc... (cached) gcc
checking whether we are using the GNU C compiler... (cached) yes
checking whether gcc accepts -g... (cached) yes
checking for gcc option to accept ISO C89... (cached) none needed
checking for g++... no
checking for c++... no
checking for gpp... no
checking for aCC... no
checking for CC... no
checking for cxx... no
checking for cc++... no
checking for cl.exe... no
checking for FCC... no
checking for KCC... no
checking for RCC... no
checking for xlC_r... no
checking for xlC... no
checking whether we are using the GNU C++ compiler... no
checking whether g++ accepts -g... no
checking how to run the C preprocessor... gcc -E
checking how to run the C++ preprocessor... /lib/cpp
configure: error: in `/usr/local/asterisk-1.4.26':
configure: error: C++ preprocessor /lib/cpp fails sanity check
See `config.log' for more details.
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[asterisk-users] Inquiry:Connect my Asterisk to external sip?
Dear All
I have an application that calls for my Asterisk sip to be connected to an
external sip server for voip routing . Please be informed that my Asterisk
sip is at @192.168.0.2 and the external sip is at @192.168.0.139 . To this
end , I modified my sip.conf extensions.conf as the followings :
Under sip.conf :
-
[general]
register = toronto:welc...@192.168.0.139/osaka
[osaka]
type=friend
secret=welcome
context=osaka_incoming
host=dynamic
disallow=all
allow=alaw
[6672019]
type=friend
host=dynamic
context=phones
Under extensions.conf :
-
[osaka_incoming]
include=local-lines
[local-lines]
exten = _XXX,n,Dial(SIP/osaka/${EXTEN})
Please find attached the log captured when making calls (the call cannot get
through) .Can you please do me favor and let me know what is wrong in my sip
configuration ?
Let me thank you in advance
log-sip
Description: Binary data
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Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @192.168.0.139. To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk sip server?
Dear All I have an application that calls for Asterisk sip configuration to be able to communicate with external sip server . My Asterisk 3.1.14 has been installed on Debian 3.1 server and the external sip server is @192.168.0.10, the same subnet as my Debian server @ 192.168.0.2 . At now , the configuration is in such a way that the call attempts reaching to my Asterisk are being routed internally , based on my Asterisk extensions.conf settings . I need to change the current configuration in such a way that the voip call attempts to be routed toward the external sip ser...@192.168.0.10 for the call routing purposes . Can you please help me how I am expected to modify my Asterisk configuration to do the job ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server
Thank you for your reply . Please find below the required data : #cat /etc/apt/sources.list #deb file:///cdrom/sarge main deb cdrom :[Debian GNU/Linux 3.1 r1a_Sarge_-Official i386 Binary-1 (20051224)] /unstable contrib main deb http://security.debian.org/stable/updates main contrib Thank you in advance On Sun, Nov 15, 2009 at 6:36 AM, Jarrod Lash jar...@fed-com.com wrote: you are running a old version of debian? what repository are you using (cat /etc/apt/sources.list)? On Sun, Nov 15, 2009 at 1:27 AM, hadi motamedi motamed...@gmail.comwrote: Sorry . I tried to install gcc but I got the following error : #apt-get update #apt-get install gcc E:Package gcc has no installation candidate Can you please do me favor and let me know why ? Thank you in advance On Sun, Nov 15, 2009 at 6:01 AM, Jarrod Lash jar...@fed-com.com wrote: apt-get update then apt-get install gcc g++ -- Jarrod Lash, jar...@fed-com.com Federated Communications www.fed-com.com Office: +1-412-357-2127 Mobile: +1-412-999-0049 Fax: +1-412-545-8368 On Sun, Nov 15, 2009 at 12:31 AM, hadi motamedi motamed...@gmail.comwrote: Dear All Please be informed that I need to install Asterisk 1.4.13 on my Debian 3.1 server . But I got the following message when trying for #./configure : error: no acceptable C compiler found in $PATH Can you please do me favor and let me know what is the problem ? Let me thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jarrod Lash, jar...@fed-com.com Federated Communications, LLC. www.fed-com.com Office: +1-412-357-2127 Mobile: +1-412-999-0049 Fax: +1-412-545-8368 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Where to download Asterisk 1.4.13 for Debian server?
Dear All Can you please do me favor and let me have the link to download the Asterisk 1.4.13 for my Debian server ? Please let me know how to install it . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server
Dear All Please be informed that I need to install Asterisk 1.4.13 on my Debian 3.1 server . But I got the following message when trying for #./configure : error: no acceptable C compiler found in $PATH Can you please do me favor and let me know what is the problem ? Let me thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server
Sorry . I tried to install gcc but I got the following error : #apt-get update #apt-get install gcc E:Package gcc has no installation candidate Can you please do me favor and let me know why ? Thank you in advance On Sun, Nov 15, 2009 at 6:01 AM, Jarrod Lash jar...@fed-com.com wrote: apt-get update then apt-get install gcc g++ -- Jarrod Lash, jar...@fed-com.com Federated Communications www.fed-com.com Office: +1-412-357-2127 Mobile: +1-412-999-0049 Fax: +1-412-545-8368 On Sun, Nov 15, 2009 at 12:31 AM, hadi motamedi motamed...@gmail.comwrote: Dear All Please be informed that I need to install Asterisk 1.4.13 on my Debian 3.1 server . But I got the following message when trying for #./configure : error: no acceptable C compiler found in $PATH Can you please do me favor and let me know what is the problem ? Let me thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:How to stop Asterisk?
Dear All Can you please do me favor and let me know how can I stop my Asterisk ? Can you please confirm if the following procedure is correct to stop it ? #/etc/init.d/asterisk stop #cd /etc/init.d #chmod asterisk Let me thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk server remote access
Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk server remote access
Thank you for your reply . But I am seeking for PPPoE remote access that fits my case here . Can you please let me know if there is any solution in this regard ? (like PPPD) On Sat, Sep 26, 2009 at 12:16 PM, Michiel van Baak mich...@vanbaak.infowrote: On 09:41, Sat 26 Sep 09, hadi motamedi wrote: Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance ssh usern...@server -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk server remote access
Thank you for your reply . Can you please let me know if there is an facility to provide PPP over E1 as my Asterisk has ISDN PRI link outwards ? I mean if any facility inside Asterisk can provide PPP over E1 for remote access via ISDN PRI link ? On Sat, Sep 26, 2009 at 11:18 AM, ravi kumar ravi...@gmail.com wrote: use Asterisk now software. You can access by IP. On Sat, Sep 26, 2009 at 2:11 PM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to convert *.wav files ?
Thank you very much for your confirmation . Excuse me , the format needs to be like the followings ? exten = s-NOANSWER,n,playback(FR1.sln) Can you please do me favor and confirm if the above is correct ? On Sat, Sep 26, 2009 at 7:42 AM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: A good way is to give try On Sat, Sep 26, 2009 at 11:41 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: yeah it can :) On Sat, Sep 26, 2009 at 11:30 AM, hadi motamedi motamed...@gmail.comwrote: Thank you for your reply . Excuse me , you mean the Asterisk can play SLN files ? Can you please confirm ? On Sat, Sep 26, 2009 at 6:57 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello Hadi In beginning i also face this problem . I solved it by converting to SLN format. You also try to convert it to sln format. this link might help you http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk On Sat, Sep 26, 2009 at 10:44 AM, hadi motamedi motamed...@gmail.com wrote: Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR3.wav FR3.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is reported as WAVE audio mono 8000 Hz) into 32 bit 44 KHz for better voice quality . Can you please help me . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Which codec to get higher download rate on dialup connection
Dear All Can you please do me favor and let me know which Asterisk codec you will prefer when you want to offer your subscribers with dialup data connection ? Let me thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk sound files
No . I don't receive any error message after converting from *.wav to *.gsm but the new announcements cannot be heared (when trying for playback). On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk sound files
Thank you . Please be informed that the *.wav files cannot be played on my Asterisk so I had to convert to *.gsm file format .I tried to convert to *.gsm by making use of sox but the new announcement cannot be heard . On Tue, Sep 8, 2009 at 7:22 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: check the file formats first if .wav is listed there and if it is, then check the translation if its activated. On Tue, Sep 8, 2009 at 2:05 PM, hadi motamedi motamed...@gmail.comwrote: No . I don't receive any error message after converting from *.wav to *.gsm but the new announcements cannot be heared (when trying for playback). On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Problem with Call Parking
Dear All I sent you a message regarding my problem with Asterisk Call Parking feature and you told me that needs to check the polycom sip.cfg file . But my Asterisk doesn't have sip.cfg file . Can you please let me know how can I overcome ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk sound files
Thank you . Please find below my original and converted sound files attributes on my Asterisk : #file FR1.wav FR1.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz #file FR1.gsm FR1.gsm: data Can you please let me know what is the problem as the sox does not generate error message when converting ? On Tue, Sep 8, 2009 at 7:57 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: I'm not quite sure but i think if you converted the file ex: file.wav using sox it should produce something like file.ulaw, file.alaw, file.gsm. Check if its there, then check the translation if you have the codec activated, it worked for me before. On Tue, Sep 8, 2009 at 2:40 PM, hadi motamedi motamed...@gmail.comwrote: Thank you . Please be informed that the *.wav files cannot be played on my Asterisk so I had to convert to *.gsm file format .I tried to convert to *.gsm by making use of sox but the new announcement cannot be heard . On Tue, Sep 8, 2009 at 7:22 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: check the file formats first if .wav is listed there and if it is, then check the translation if its activated. On Tue, Sep 8, 2009 at 2:05 PM, hadi motamedi motamed...@gmail.comwrote: No . I don't receive any error message after converting from *.wav to *.gsm but the new announcements cannot be heared (when trying for playback). On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk sound files
Please find below the error message that I am receiving on my Asterisk : -- Executing [s-noans...@macro-dialuser:4] Playback(Zap/95-1, FR1) in new stack [Sep 7 11:11:34] WARNING[7624]: format_wav.c:140 check_header: Not a wav file 6 [Sep 7 11:11:34] WARNING[7624]: file.c:316 fn_wrapper: Unable to open format wav WARNING[7624]: file.c:866 ast_streamfile: Unable to open FR1 (format 0x48 (alaw|slin)): WARNING[7624]: app_playback.c:437 playback_exec: ast_streamfile failed on Zap/95-1 for FR1 On Tue, Sep 8, 2009 at 9:34 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: is there an error on the asterisk cli when you're playing the sound file? On Tue, Sep 8, 2009 at 4:19 PM, hadi motamedi motamed...@gmail.comwrote: Thank you . Please find below my original and converted sound files attributes on my Asterisk : #file FR1.wav FR1.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz #file FR1.gsm FR1.gsm: data Can you please let me know what is the problem as the sox does not generate error message when converting ? On Tue, Sep 8, 2009 at 7:57 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: I'm not quite sure but i think if you converted the file ex: file.wav using sox it should produce something like file.ulaw, file.alaw, file.gsm. Check if its there, then check the translation if you have the codec activated, it worked for me before. On Tue, Sep 8, 2009 at 2:40 PM, hadi motamedi motamed...@gmail.comwrote: Thank you . Please be informed that the *.wav files cannot be played on my Asterisk so I had to convert to *.gsm file format .I tried to convert to *.gsm by making use of sox but the new announcement cannot be heard . On Tue, Sep 8, 2009 at 7:22 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: check the file formats first if .wav is listed there and if it is, then check the translation if its activated. On Tue, Sep 8, 2009 at 2:05 PM, hadi motamedi motamed...@gmail.comwrote: No . I don't receive any error message after converting from *.wav to *.gsm but the new announcements cannot be heared (when trying for playback). On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.com wrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Thank you for your message . But I tried to find it on my server , as the followings : #find / -name sip.cfg -print But it didn't return any result . Can you please let me know where can I find it ? On Tue, Sep 8, 2009 at 2:58 PM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 8 Sep 2009, hadi motamedi wrote: I sent you a message regarding my problem with Asterisk Call Parking feature and you told me that needs to check the polycom sip.cfg file . But my Asterisk doesn't have sip.cfg file . Can you please let me know how can I overcome ? sip.cfg is not an Asterisk file. sip.cfg should be in the directory the phone downloads it's configuration from. Typically, /tftpboot/ on a tftp server. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Sorry , I checked on my Asterisk pbx and there is no sip.cfg file on it . Can you please let me know how can I make my Asterisk Call Parking as functional ? On Tue, Sep 1, 2009 at 6:23 AM, Lee, John (Sydney) john@compuware.comwrote: Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Yes, I am referring to the Polycom sip.cfg and not the sip.cfg in Asterisk. Somethere down in sip.cfg, there is a line that looks like this: digitmap dialplan.digitmap=#700| ... Basically, Polycom will scan your input to see when it will pass all the keystrokes to Asterisk. In above, if it detects that you have entered #700, it will automatically send it to Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk sound files
Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi FR1.gsm Description: Binary data FR3.gsm Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Thank you for your reply . Do you mean my Asterisk extensions.conf must contain a line like the followings ? include = parkedcalls If so , can you please let me know where I have to put this line in my extensions.conf ? Thank you in advance Regards H.Motamedi On Thu, Sep 3, 2009 at 5:26 AM, Stephen Davies stephen.l.dav...@gmail.comwrote: In any event, the real problem is probably that you forgot to 'include = parkedcalls' in your dialplan. Steve On 9/2/09, Lyle Giese l...@lcrcomputer.net wrote: And now that the whole world of Asterisk has your sip user ids and passwords, you should change all of the passwords that are in that file and yes, change the passwords in all your phones. Lyle Giese LCR Computer Services, Inc. hadi motamedi wrote: Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.com mailto:john@compuware.com wrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Tuesday, 1 September 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inquiry:Problem with Call Parking Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from my mobile device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Problem with VoiceMail
Dear All
Can you please do me favor and let me know what is my problem with my
Asterisk VoiceMail configuration as it doesn't work correctly in my case ?
Please find below that part of my extensions.conf that I intend to make use
of voice mail for No Answer reply :
[line-incoming]
exten = _XXX,1,macro(dialuser,SIP/${EXTEN},${EXTEN})
[macro-dialuser]
exten = s,1,dial(${ARG1},38,r)
exten = s,n,noop(PM: Dial ended !!)
exten = s,n,noop(${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,answer
exten = s-NOANSWER,n,wait(4)
exten = s-NOANSWER,n,SayDigits(${ARG2})
exten = s-NOANSWER,n,playback(vm-isunavail)
exten = s-NOANSWER,n,VoiceMail(u${MACRO_EXTEN})
exten = s-NOANSWER,n,hangup()
As you see , I intend to redirect the calling party to the called party
voice mailbox if he doesn't answer the call (that will be set at the number
the same as his extension number) but it doesn't get through. Can you please
let me know what is wrong in our case ?
Regards
H.Motamedi
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Re: [asterisk-users] Inquiry:Problem with VoiceMail
Thank you for your reply . The part of the code that I sent you is for the
case the called party didn't answer . I want to get the calling party
message into the called party voice mail box . Please help me to correct my
code .
Regards
H.Motamedi
On Tue, Sep 1, 2009 at 8:52 AM, Matt Riddell li...@venturevoip.com wrote:
On 1/09/09 6:14 PM, hadi motamedi wrote:
exten = s,n,noop(${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1)
As you see , I intend to redirect the calling party to the called party
voice mailbox if he doesn't answer the call (that will be set at the
number the same as his extension number) but it doesn't get through. Can
you please let me know what is wrong in our case ?
In the Asterisk console, what does it say the dialstatus is?
--
Cheers,
Matt Riddell
Director
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http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
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Re: [asterisk-users] Inquiry:How to hide Caller Id
Thank you for your reply . Yes , he is seeing his own number on his phone upon going off hook and before dialing any number . Can you please do me favor and confirm if it is not a feature of Asterisk that I can disable it ? Regards H.Motamedi On Mon, Aug 31, 2009 at 6:53 AM, Matt Riddell li...@venturevoip.com wrote: On 31/08/09 5:49 PM, hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Hi, If he is seeing his own number on his display before he has dialed any numbers then it is probably a feature of the phone - in which case you need to disable it there. If you're talking about an incoming call then it's different. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
Thank you for your reply . I really don't want the user to know the phone's local number . Can you please do me favor and propose one of the available phones that doesn't have this feature ? Regards H.Motamedi On Mon, Aug 31, 2009 at 7:12 AM, Cary Fitch ca...@usawide.net wrote: That doesn’t happen on all phones. Either find a way to block that “feature” on the phone, or change phones for that location. I assume you don’t want the user to know that phone’s local number. Cary Fitch -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *hadi motamedi *Sent:* Monday, August 31, 2009 1:09 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Inquiry:How to hide Caller Id Thank you for your reply . Yes , he is seeing his own number on his phone upon going off hook and before dialing any number . Can you please do me favor and confirm if it is not a feature of Asterisk that I can disable it ? Regards H.Motamedi On Mon, Aug 31, 2009 at 6:53 AM, Matt Riddell li...@venturevoip.com wrote: On 31/08/09 5:49 PM, hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Hi, If he is seeing his own number on his display before he has dialed any numbers then it is probably a feature of the phone - in which case you need to disable it there. If you're talking about an incoming call then it's different. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
Sorry for mis-typing in phone type . Please be informed that the current phone type our subscribers are using is TP6000 ones . On Mon, Aug 31, 2009 at 7:24 AM, Paul Hales pdha...@optusnet.com.au wrote: I couldn't find any information on this brand of phone on the internet at all. PaulH hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Looking forward your reply Regards H.Motamedi On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell li...@venturevoip.com mailto:li...@venturevoip.com wrote: On 31/08/09 5:24 PM, hadi motamedi wrote: Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . You're not really giving enough information. Who sees the number? Where do they see it? What type of phone? What is a subs? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Problem with Call Parking
Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi features.conf Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.comwrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Tuesday, 1 September 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inquiry:Problem with Call Parking Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users sip.conf Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:How to hide Caller Id
Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Looking forward your reply Regards H.Motamedi On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell li...@venturevoip.com wrote: On 31/08/09 5:24 PM, hadi motamedi wrote: Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . You're not really giving enough information. Who sees the number? Where do they see it? What type of phone? What is a subs? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
Sorry . I meant subscriber . On Mon, Aug 31, 2009 at 6:31 AM, Paul Hales pdha...@optusnet.com.au wrote: Matt Riddell wrote: What is a subs? A submarine. I think. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk Inter digit delay
Dear David I appreciate your reply . Pleae find attached our current extensions.conf file . Can you please do me favor and let me know where I am expected to put your proposed line for defining the timeout param ? Regards H.Motamedi On Fri, Jul 31, 2009 at 3:16 AM, David Backeberg dbackeb...@gmail.comwrote: On Thu, Jul 30, 2009 at 1:19 AM, hadi motamedimotamed...@gmail.com wrote: Thank you very much for your reply . But please be informed that our current line-outgoing route is being configured as the followings (in extensions.conf): Set(TIMEOUT(digit)=timeout) There's definitely more to your dialplan than the sample you provided. You need to add the timeout in there. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users extensions.conf Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk supporting hash (#) key
Dear All Please be informed that we have an application for our subs to be able to dial #21 to reach IN services . Can you please let us know how we can support for this as it seems that the Asterisk does not support for the hash # key as an valid extension to be dialed by the user ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry : Asterisk hash key
Dear All Can you please let us know how to configure Asterisk to recognize extensions starting with the hash key ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk Inter digit delay
Dear David I mean when the subs dials the digits with some delay between entering the digits sequentially . At our current case , the Asterisk will wait about 2 seconds to see if another digit will be dialed or not and then he will route the dialed digits according to the pre-defined routing table or he will play the appropriate announcements . We are expected to increase this inter digit delay to say 4 seconds . Please let us know how we can increase this parameter in our Asterisk configuration files . Regards H.Motamedi On Thu, Jul 30, 2009 at 3:19 AM, David Backeberg dbackeb...@gmail.comwrote: On Tue, Jul 28, 2009 at 1:01 AM, hadi motamedimotamed...@gmail.com wrote: Dear All Can you please let us know how we can modify our Asterisk inter digit delay ? Actually , our subs dials his intended numbers with some delay in between entering the digits sequentially . It seems that our Asterisk pbx will wait for about 2 seconds and if no extra digits are to be entered then he will decide on routing the dialed number or play the related anouncement . For our current application , it seems that this amount of delay is a little bit small and so please let us know how we can incraese this amount of delay to say 4 seconds . Delay between what? Is this a phone being dialed with a dial tone? Is this an IVR prompt timeout? If you do a Read(), do a longer timeout. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk Inter digit delay
Thank you very much for your reply . But please be informed that our current
line-outgoing route is being configured as the followings (in
extensions.conf):
[line-outgoing]
exten = _X.,1,macro(dialuser,Zap/g1/${EXTEN},${EXTEN})
As you see , it is trying to consider the dialed number as an whole packet
(but not based on one-by-one digit basis) . Can you please do us favor and
let us know how we can get benefit of your proposed DigitTimeout command to
fulfill the job ?
Regards
H.Motamedi
On Thu, Jul 30, 2009 at 5:28 AM, Marc Charbonneau
timebandit...@gmail.comwrote:
I mean when the subs dials the digits with some delay between entering
the
digits sequentially . At our current case , the Asterisk will wait about
2
seconds to see if another digit will be dialed or not and then he will
route
the dialed digits according to the pre-defined routing table or he will
play
the appropriate announcements . We are expected to increase this inter
digit
delay to say 4 seconds . Please let us know how we can increase this
parameter in our Asterisk configuration files .
see http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout
also, I suggest you read this book : http://astbook.asteriskdocs.org/
hth
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[asterisk-users] Inquiry:Asterisk * character dialing for IN service
Dear All
Can you please let us know how we can modify our outgoing extension routing
such that our subs can dial as *21 for reaching to IN services . Please
find below our current item for outgoing dialing , as the followings :
[line-outgoing]
exten = _X.,1,macro(dialuser,Zap/g1/${EXTEN},${EXTEN})
At now , as you see , we are sending all of the subs dialed digits to the
peer side . Please let us know how we can modify the above line to include
for the case that our subs dials as *21 and we still want to route it to
the peer side as well .
Regards
H.Motamedi
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[asterisk-users] Inquiry abount Asterisk extensions.conf
Dear All Regarding our opened case , can you please confirm if our attached extensions.conf file can fullfil the needs of sending the subs dialed digits one-by-one instead of sending it as an whole packet ? Regards H.Motamedi extensions.conf Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk Inter digit delay
Dear All Can you please let us know how we can modify our Asterisk inter digit delay ? Actually , our subs dials his intended numbers with some delay in between entering the digits sequentially . It seems that our Asterisk pbx will wait for about 2 seconds and if no extra digits are to be entered then he will decide on routing the dialed number or play the related anouncement . For our current application , it seems that this amount of delay is a little bit small and so please let us know how we can incraese this amount of delay to say 4 seconds . Looking for your reply Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk pbx announcements
Dear All It seems that our Asterisk pbx announcement files are being stored inside the /var/lib/asterisk/sounds folder . Can you please let us know what is the appropriate program to open and hear them on an MS Windows client ? (e.g. pbx-invalid.gsm) Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry abount Asterisk extensions.conf
Dear John The peer switch is Huawei switch and we need this functionality as we support ISDN PRI but it supports for V5 interface . So another softswitch is functioning between us . The peer side expects to receive the subscriber dialed digits one-by-one as he sees us as an access equipment (not just an PBX) . Regards H.Motamedi On Wed, Jul 22, 2009 at 12:53 PM, John Novack jnov...@stromberg-carlson.org wrote: Curious - Why? What is the peer switch and why does it have this requirement? John Novack hadi motamedi wrote: Dear All Can you please let us know how we can modify our Asterisk extensions.conf file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as 665 so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one 665 packet . Your reply is very welcome Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry abount Asterisk extensions.conf
Dear Leif Can you please provide us with more details on this Overlap Dialing phillosophy ? Regards H.Motamedi On Wed, Jul 22, 2009 at 1:15 PM, Leif Madsen leif.mad...@asteriskdocs.orgwrote: John Novack wrote: Can you please let us know how we can modify our Asterisk extensions.conf file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as 665 so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one 665 packet . Curious - Why? What is the peer switch and why does it have this requirement? That's a funny way of answering the question :) I *think* what he wants is overlap dialing. Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry abount Asterisk extensions.conf
Dear All Can you please let us know how we can modify our Asterisk extensions.conf file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as 665 so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one 665 packet . Your reply is very welcome Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:How to convert *.wav files ?
-> [asterisk-users] Inquiry:How to convert *.wav files ? asterisk-users -- Thread -- -- Date -- [asterisk-users] Inquiry:How to convert *.wav files ? hadi motamedi Re: [asterisk-users] Inquiry:How to convert *.wav files ? ABBAS SHAKEEL Re: [asterisk-users] Inquiry:How to convert *.wav files ? hadi motamedi Re: [asterisk-users] Inquiry:How to convert *.wav files ? ABBAS SHAKEEL Re: [asterisk-users] Inquiry:How to convert *.wav files ? ABBAS SHAKEEL Re: [asterisk-users] Inquiry:How to convert *.wav files ? hadi motamedi Re: [asterisk-users] Inquiry:How to convert *.wav files ? Stanisław Pitucha Re: [asterisk-users] Inquiry:How to convert *.wav files ? ravi kumar Re: [asterisk-users] Inquiry:How to convert *.wav files ? ABBAS SHAKEEL Reply via email to
[asterisk-users] Inquiry:How to convert *.wav files ?
-> Re: [asterisk-users] Inquiry:How to convert *.wav files ? asterisk-users -- Thread -- -- Date -- [asterisk-users] Inquiry:How to convert *.wav files ? hadi motamedi Re: [asterisk-users] Inquiry:How to convert *.wav files ? ABBAS SHAKEEL Re: [asterisk-users] Inquiry:How to convert *.wav files ? hadi motamedi Re: [asterisk-users] Inquiry:How to convert *.wav files ? ABBAS SHAKEEL Re: [asterisk-users] Inquiry:How to convert *.wav files ? ABBAS SHAKEEL Re: [asterisk-users] Inquiry:How to convert *.wav files ? hadi motamedi Re: [asterisk-users] Inquiry:How to convert *.wav files ? Stanisław Pitucha Re: [asterisk-users] Inquiry:How to convert *.wav files ? ravi kumar Re: [asterisk-users] Inquiry:How to convert *.wav files ? ABBAS SHAKEEL Reply via email to
