RE: [Asterisk-Users] Best bet ... IAX vs SIP
Pentium 4 2.8 - 2 gigs ram - raided 80 gig drives - 1 T1 card -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nir Simionovich Sent: Sunday, June 12, 2005 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best bet ... IAX vs SIP Well, I think you are asking the wrong question here, I think the proper question would be: In a 20 extension iPBX environment, what combination of signaling and codec would provide the best performace on a hardware of [specify your hardware here]? Nir S mr. barker wrote: > As the topic states. > > > > In a 20 phone PBX enviroment. I am wondering which would have the > greater server and router load. > > > > What do you use ? > > > > Thank you. > > > >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best bet ... IAX vs SIP
As the topic states. In a 20 phone PBX enviroment. I am wondering which would have the greater server and router load. What do you use ? Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiPSupply Dot Com
. Snip It is sad to hear that you will not be purchasing from us. I do not understand though, why we owe you an explanation for our toll free number being down. ^^ You are right you don't owe any explanation at all for your numbers being down. It was your Toll Free and Your Toll number! Not just your Toll Free. For me personally once I find a company to deal with I usually stick with them. I have ordered in an excess of $3000.00 from your company. When I did call last week I tried on numerous ocassions and had the same result. I even call the information and there was no listing for B2 Technologies nor VoipSupply (this doesn't mean much though as you must subscribe to the listing service). What strikes me very odd is that I tried to call the numbers from work and from home same results no longer in service. I also called through the my VOIP provider (which routes down to the US) and the local teleco here. I wanted to make sure that something was not up so I asked my brother to try placing a call (he lives in a different province) and he had the same no long in service. This is not just one isolated incident involving one call! I was going to post something last week but decided to not as I wanted to see how your response was in the community. As I predicated there would be someone posting a something to the effect about "how is VoipSupply to deal with" then followed by people saying that the service is reliable ... etc. Now I may be just a bit over cautious when it comes to dealing with internet based businesses because of being burnt before along with 1000's of others. This is just my 2cents. Lastly, we do charge for technical support. We are hear to help, but the low margins on ATA's etc certainly does not leave us room to give away free support. All of you that are ITSP's know exactly what I am talking about. If you order something, and you can't get it to work, you can pay for us to make it work for you. If you order the wrong product, then that is your mistake not ours. There is an open invite to all to call or email me at any time to discuss or business. Constructive criticism is always welcomed. Thank you all for business and we look for more in the future! Garrett Smith VoIPSupply.com [EMAIL PROTECTED] 716-250-3408 Direct > mr. barker wrote: > >> I tried calling their toll free number and toll number last week in >> the morning and afternoon and was handed a recording saying this >> number is no longer in service. The web site was up but there was no >> message on the site as to why the phone numbers were not working. >> >> I just called the number now and it is working. >> >> Being around the internet for a quite a long time this gives me an >> uneasy feeling. I have seen company's start to go under and pull the >> plug when they get into financial trouble(not being able to pay the >> bills) and run with the customers money. I have had this happen to me >> on 2 occasions. Just the woes of doing business on the net. >> >> Being in Canada it makes it very difficult to find companies that will >> ship COD from the US. If I was to order I would only order COD from >> now on from VoipSupply. >> >> I have ordered product from VoipSupply and received the product. I >> will not be ordering more product do to this outage of the phones with >> no explanation. >> >> Just my 2cents. >> > Maybe the tollfree provider was responsible for the outage and maybe it > only affected service from Canada. > > They accept credit cards and paypal. I believe you would have some > recourse if they ran with your money. > > I quit shipping anything COD to anywhere a few years ago. If the > customer refuses delivery the vendor loses money. When UPS instituted a > policy of not handling cash payment for COD, I quit for good. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiPSupply Dot Com
I tried calling their toll free number and toll number last week in the morning and afternoon and was handed a recording saying this number is no longer in service. The web site was up but there was no message on the site as to why the phone numbers were not working. I just called the number now and it is working. Being around the internet for a quite a long time this gives me an uneasy feeling. I have seen company’s start to go under and pull the plug when they get into financial trouble(not being able to pay the bills) and run with the customers money. I have had this happen to me on 2 occasions. Just the woes of doing business on the net. Being in Canada it makes it very difficult to find companies that will ship COD from the US. If I was to order I would only order COD from now on from VoipSupply. I have ordered product from VoipSupply and received the product. I will not be ordering more product do to this outage of the phones with no explanation. Just my 2cents. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help Please Multiple Users for Broadvoice
I would like to be able to have multiple users (the wife and kids) to be able to access the Broadvoice account at the same. No complaining that way from them J. I seen someones configuration in the group here but now I can’t find it (lost my glasses). If someone could post theirs’s or the shortcut that would be great. Thanks for your help. “Dad she’s on the phone again !” ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
Thank you to both Chris and Tim I could not get my head around this .. after seeing the examples it now makes sense what needs to be done. I will give both a whirl tonight. I do like the RSA key idea. One question is this, will I need multiple accounts on the Static IP machines so the Dynamic machine has the ability to make more then one concurrent SIP call through the Static IP machine ? If I could get the Static IP box to go through the my SMC router it would be great. I tried opening the ports. 5060udp/tcp, 1-2udp/tcp. Tried even setting the machine in the DMZ zone. I think the VOIP provider just has problems translating through the NAT or something. The linux box is running [EMAIL PROTECTED] no firewall setting that I know of. To much of a Newbie at linux .. lol and I have been at it for almost 1 year now and still have s much to learn. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Sent: Thursday, May 05, 2005 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out I haven't gotten to keys yet. The documentation out there doesn't seem to be very good. Chris - Original Message - From: "Tim Pushor" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, May 05, 2005 4:06 PM Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out > Personally, if I owned both boxes and had full control of the dialplan > on both, I'd stay away from passwords. (but be careful what I say, I'm a > hack) > > I have a bunch of boxes connected together via IAX and authenticating > via RSA. The entries in iax.conf are simple, and dialing across the > connection is simple (no passwords in the dialplan) (thanks again Rich > for taking the time). > > Tim > > Here is a sample of iax.conf entries on machine a: > > [machineb] > type=user > host=machineb.internal.net > auth=rsa > inkeys=machineb > username=machineb > context=inbound > > [machineb] > type=peer > host=machineb.internal.net > auth=rsa > outkey=machinea > username=machinea > > And an example dialplan entry to dial an extention on machineb (in the > inbound context): > > exten => 333,1,Dial(IAX2/machineb/333) > > And on machinea, the opposite of machineb: > > [machinea] > type=user > host=machinea.internal.net > auth=rsa > inkeys=machinea > username=machinea > context=inbound > > [machinea] > type=peer > host=machinea.internal.net > auth=rsa > outkey=machineb > username=machineb > > To generate the keys: > > on machinea: > > astgenkey -n machinea > mv machinea.* /var/lib/asterisk/keys > > copy machinea.pub to machineb's /var/lib/asterisk/keys > > on machineb: > > astgenkey -n machineb > mv machineb.* /var/lib/asterisk/keys > > copy machineb.pub to machinea's /var/lib/asterisk/keys > > > Chris wrote: > > >I have something similar. Both of my servers are behind a firewall and NAT. You will need to allow UDP 4569 through the firewall for IAX2. If you have NAT you will need to redirect 4569 to the internal server. > > > >I would suggest using AMP and then looking at IAX_ADDITIONAL.CONF to see how it's done. You can modify the IAX.CONf because I don't believe AMP rewrites that file. > > > >I think the user and passwords are required. I would suggest using a strong password or someone may decide to make a few phone calls. After this you will need the routing in Extensions.conf to allow calls to be made on this trunk. > > > >Asterisk will handle the SIP > IAX.All my clients are SIP and they have no trouble going over a IAX trunk to other SIP devices on the other server. > > > >This is what my IAX_ADDITIONAL.CONF looks like > > > >SiteA - Dynamic IP > >-- > >[boxb-peer] > >username=boxa-user > >type=peer > >trunk=yes > >secret=mypassword > >host=thehost.dyndns.org > > > >[boxb-user] > >type=user > >secret=mypassword2 > >host=thehost.dyndns.org > >context=from-internal > > > >--- > >Site b - Static IP > > > > > >[boxa-peer] > >username=boxb-user > >type=peer > >trunk=yes > >secret=mypassword2 > >host=xxx.xxx.xxx.xxx > > > >[boxa-user] > >type=user > >secret=mypassword > >host=xxx.xxx.xxx.xxx > >context=from-internal > > > > > >Regards, > > > >Chris > > > > > >- Original Message - > >F
RE: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
Yes trying to connect to boxes together. One sits outside the internal firewall and is on the inside. I am using AMP. However I can just put whatever I need in the custom.conf sections. The users agents are SIP .. can SIP call go over a IAX trunk ? if so great. To create the trunk do I need to use a users name and password ? or ? I need to have the *box that is behind the firewall to be able to place a call out through the *box that has a public ip. Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Sent: Thursday, May 05, 2005 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out I am not sure what you are trying to do.I have created an IAX2 trunk between the servers over an internet connection. Then all you have to do is put in call routing on the trunks to forward the call to the right place. Are you using AMP or trying to do it manually. I found everything a little confusing as well, but it is simple now that I understand it. Chris - Original Message - From: "mr. barker" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Thursday, May 05, 2005 4:43 AM Subject: [Asterisk-Users] Connecting 2 * Together-Pulling hair out > > > > > _ > > Subject: [Asterisk-Users] Connecting 2 * Together-Pulling hair out > > > > I have read the docs on connecting 2* together but am unsure of a few things > > > > Do I need a different account for each number that will be called from one > box to the other ? ie. Do I set up a user account on one and then have the > other box log into that account when it whats to make a call ? > > > > I have 2 asterisk boxes and only one of them has the ability to access a > VoipAccount and PSTN connections.(*box 1). The other holds the SIP > extensions for the internal SIP users/exten(*box2) > > I would like to be able to have the box with the Sip UA(*box2) on it to be > able to place a call using the box that has the VoipAccount and PSTN > connection. I am able to make multiple UA calls on the VoipAccount and 3 on > the PSTN lines (only have 3 lines coming in). I can get it to work if I > create a user exten on *box1 and map a trunk(which is really only an exten) > using the user/password login to that exten from *box2. However when I try > to place a second call when the VOIP line is in use it gives me error ( > basically saying can't use the trunk because it is in use) I would like to > be able to have this exten/trunk to be able to use multiple connections on > it. > > > > There must be an easier way to do this I am just not sure how. I looked at > creating IAX trunks but still come up with the Trunk is really an Exten > name/password . > > > > Any help would be appreciated. (my brain is boiling eggs) > > > > Thank you. > > > > > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting 2 * Together-Pulling hair out
Subject: [Asterisk-Users] Connecting 2 * Together-Pulling hair out I have read the docs on connecting 2* together but am unsure of a few things Do I need a different account for each number that will be called from one box to the other ? ie. Do I set up a user account on one and then have the other box log into that account when it whats to make a call ? I have 2 asterisk boxes and only one of them has the ability to access a VoipAccount and PSTN connections.(*box 1). The other holds the SIP extensions for the internal SIP users/exten(*box2) I would like to be able to have the box with the Sip UA(*box2) on it to be able to place a call using the box that has the VoipAccount and PSTN connection. I am able to make multiple UA calls on the VoipAccount and 3 on the PSTN lines (only have 3 lines coming in). I can get it to work if I create a user exten on *box1 and map a trunk(which is really only an exten) using the user/password login to that exten from *box2. However when I try to place a second call when the VOIP line is in use it gives me error ( basically saying can’t use the trunk because it is in use) I would like to be able to have this exten/trunk to be able to use multiple connections on it. There must be an easier way to do this I am just not sure how. I looked at creating IAX trunks but still come up with the Trunk is really an Exten name/password . Any help would be appreciated. (my brain is boiling eggs) Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting 2 * Together-Pulling hair out
I have read the docs on connecting 2* together but am unsure of a few things Do I need a different account for each number that will be called from one box to the other ? ie. Do I set up a user account on one and then have the other box log into that account when it whats to make a call ? I have 2 asterisk boxes and only one of them has the ability to access a VoipAccount and PSTN connections.(*box 1). The other holds the SIP extensions for the internal SIP users/exten(*box2) I would like to be able to have the box with the Sip UA(*box2) on it to be able to place a call using the box that has the VoipAccount and PSTN connection. I am able to make multiple UA calls on the VoipAccount and 3 on the PSTN lines (only have 3 lines coming in). I can get it to work if I create a user exten on *box1 and map a trunk(which is really only an exten) using the user/password login to that exten from *box2. However when I try to place a second call when the VOIP line is in use it gives me error ( basically saying can’t use the trunk because it is in use) I would like to be able to have this exten/trunk to be able to use multiple connections on it. There must be an easier way to do this I am just not sure how. I looked at creating IAX trunks but still come up with the Trunk is really an Exten name/password . Any help would be appreciated. (my brain is boiling eggs) Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Centos - Hylafax Install
Has anyone tried to install Hylafax on Centos ? If so is there an rpm .. or what was your compiling procedure ? Thanks in return ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is this normal - Long time to make call - What is your average with your Hardware?
Hardware – Pentium 1.4 Gig – 1 Meg ram – 1 FXO100 Card – Sipura 2000 – Local Network Router SMC –Codec 711 – Asterisk @ home (lastest) On average it take almost 10 – 13 Secs to make an outbound call to a local number. Is this a normal time ? Is there something that can be done to cut this time down? Is the FXO100 the problem ? ie. Modem card ? It takes a little less time to make a Long distance call going to a 3rd party termination service. Thank you for any input. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Who is willing to help an Asterisk newby?
You can turn off the amount of logging in the log.conf setting. As far as the registration goes .. that would be under your Sipura Settings. You may only want to reduce this to 60 sec registration .. I find that any longer sometime effects longevity of server to find you in the route. Only my 2cents ... and I am no expert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolf N. Paul Sent: Wednesday, April 13, 2005 8:22 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Who is willing to help an Asterisk newby? As of last night, I have a working Asterisk system, courtesty of "[EMAIL PROTECTED]". Now comes the need to iron out the wrinkles and fine-tune my setup. Who would be willing for me to shoot questions at him/her which would just annoy the list if I brought them here? Here is my setup: P3/450Mhz, 256MB RAM, 80GB Disk (Compaq Deskpro EN), 1 X100P OEM FXO connected to PSTN (Telekom Austria) 1 Sipura 2000 connected to a Siemens Gigaset DECT/GAP Cordless unit Multiple X-lite Softphones Broadvoice configured as a "trunk" FWD configured as a "trunk" Here's one of my first questions: The system seems to be re-registering with Broadvoice every 20 seconds or so. Things work, but this seems to be an awful lot of unnecessary activity both on the network, and in the logfile. Here is how it manifests in the logfile: > Apr 13 15:18:44 DEBUG[28010]: Registration successful > Apr 13 15:18:44 DEBUG[28010]: Cancelling timeout 13449 > Apr 13 15:19:00 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 > Apr 13 15:19:00 DEBUG[28010]: Target address 147.135.4.128 is not > local, substituting externip > Apr 13 15:19:00 DEBUG[28010]: Scheduled a registration timeout # 13452 > Apr 13 15:19:00 DEBUG[28010]: Stopping retransmission on > '[EMAIL PROTECTED]' of Request 144: Found > Apr 13 15:19:00 DEBUG[28010]: Registration successful > Apr 13 15:19:00 DEBUG[28010]: Cancelling timeout 13452 > Apr 13 15:19:16 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 > Apr 13 15:19:16 DEBUG[28010]: Target address 147.135.4.128 is not > local, substituting externip > Apr 13 15:19:16 DEBUG[28010]: Scheduled a registration timeout # 13455 > Apr 13 15:19:16 DEBUG[28010]: Stopping retransmission on > '[EMAIL PROTECTED]' of Request 145: Found > Apr 13 15:19:16 DEBUG[28010]: Registration successful > Apr 13 15:19:16 DEBUG[28010]: Cancelling timeout 13455 > Apr 13 15:19:32 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 > Apr 13 15:19:32 DEBUG[28010]: Target address 147.135.4.128 is not > local, substituting externip > Apr 13 15:19:32 DEBUG[28010]: Scheduled a registration timeout # 13458 > Apr 13 15:19:32 DEBUG[28010]: Stopping retransmission on > '[EMAIL PROTECTED]' of Request 146: Found > Apr 13 15:19:32 DEBUG[28010]: Registration successful > Apr 13 15:19:32 DEBUG[28010]: Cancelling timeout 13458 > Apr 13 15:19:42 DEBUG[28010]: Manager received command 'Command' > Apr 13 15:19:42 DEBUG[28010]: Manager received command 'Command' > Apr 13 15:19:49 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 > Apr 13 15:19:49 DEBUG[28010]: Target address 147.135.4.128 is not > local, substituting externip > Apr 13 15:19:49 DEBUG[28010]: Scheduled a registration timeout # 13461 > Apr 13 15:19:49 DEBUG[28010]: Stopping retransmission on > '[EMAIL PROTECTED]' of Request 147: Found > Apr 13 15:19:49 DEBUG[28010]: Registration successful > Apr 13 15:19:49 DEBUG[28010]: Cancelling timeout 13461 > Apr 13 15:20:05 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 > Apr 13 15:20:05 DEBUG[28010]: Target address 147.135.4.128 is not > local, substituting externip > Apr 13 15:20:05 DEBUG[28010]: Scheduled a registration timeout # 13464 > Apr 13 15:20:05 DEBUG[28010]: Stopping retransmission on > '[EMAIL PROTECTED]' of Request 148: Found > Apr 13 15:20:05 DEBUG[28010]: Registration successful ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about Routing Order in .conf files
The question is in the
logical route that asterisk takes when reading and executing the scripts.
Please see the (?) questions beside the lines.
The goal is not to comment
the lines "exten => snip" in the [ext-local] everytime that I make
a change using the AMP GUI. Also it would be nice to be able to give priority
to the *_custom.conf if possible.
Thank you in return.
"Extensions_additional.conf"
[aa_1]
include => aa_1-custom
exten =>
1,1,Goto(ext-local,7726258,1) ;
exten =>
2,1,Goto(ext-local,7726259,1) ; this take the call to the
[ext-local]
exten =>
3,1,Goto(ext-local,7726257,1) ;
exten =>
fax,1,Goto(ext-fax,in_fax,1) ;
[ext-local]
include =>
ext-local-custom ; ? should this not be held in priority first over any of the
contents in [ext-local] ?
exten =>
7726257,1,Macro(exten-vm,[EMAIL PROTECTED],7726257)
exten =>
7726258,1,Macro(exten-vm,[EMAIL PROTECTED],7726258)
? I am able to
"monitor" the call if I comment the line out ? as it then seems to go
to the [ext-local-custom] located in the "extentions_custom.conf" ?
;exten =>
7726259,1,Macro(exten-vm,[EMAIL PROTECTED],7726259)
;exten =>
7726259,1,SetVar(CALLFILENAME=${CALLERIDNUM})
;exten =>
7726259,2,SetVar(CALLTIME=${DATETIME})
;exten =>
7726259,3,SetVar(CALLPATH=/var/www/html/monitor/7726259)
;exten =>
7726259,4,Monitor(wav,${CALLPATH}/${CALLTIME}-${CALLFILENAME},m)
;exten =>
7726259,5,DIAL(SIP/7726259,15,t) ;exten =>
7726259,6,Macro(exten-vm,[EMAIL PROTECTED],7726259)
exten =>
9873022,1,Macro(exten-vm,[EMAIL PROTECTED],9873022)
"extentions_custom.conf"
[ext-local-custom]
;test to see if this stays
exten =>
7726259,1,SetVar(CALLFILENAME=${CALLERIDNUM})
exten =>
7726259,2,SetVar(CALLTIME=${DATETIME})
exten =>
7726259,3,SetVar(CALLPATH=/var/www/html/monitor/7726259)
exten =>
7726259,4,Monitor(wav,${CALLPATH}/${CALLTIME}-${CALLFILENAME},m)
exten =>
7726259,5,DIAL(SIP/7726259,15,t) exten =>
7726259,6,Macro(exten-vm,[EMAIL PROTECTED],7726259)
___
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[Asterisk-Users] Script routing Logic Question in .conf files
The question is in the logical route that asterisk takes when reading and
executing the scripts. Please see the (?) questions beside the lines.
The goal is not to comment the lines "exten => snip" in the [ext-local]
everytime that I make a change using the AMP GUI. Also it would be nice to
be able to give priority to the *_custom.conf if possible.
"Extensions_additional.conf"
[aa_1]
include => aa_1-custom
exten => 1,1,Goto(ext-local,7726258,1) ;
exten => 2,1,Goto(ext-local,7726259,1) ; this take the call to the
[ext-local]
exten => 3,1,Goto(ext-local,7726257,1) ;
exten => fax,1,Goto(ext-fax,in_fax,1) ;
[ext-local]
include => ext-local-custom ; ? should this not be held in priority first
over any of the contents in [ext-local] ?
exten => 7726257,1,Macro(exten-vm,[EMAIL PROTECTED],7726257)
exten => 7726258,1,Macro(exten-vm,[EMAIL PROTECTED],7726258)
? I am able to "monitor" the call if I comment the line out ? as it then
seems to go to the [ext-local-custom] located in the
"extentions_custom.conf" ?
;exten => 7726259,1,Macro(exten-vm,[EMAIL PROTECTED],7726259)
;exten => 7726259,1,SetVar(CALLFILENAME=${CALLERIDNUM})
;exten => 7726259,2,SetVar(CALLTIME=${DATETIME})
;exten => 7726259,3,SetVar(CALLPATH=/var/www/html/monitor/7726259)
;exten => 7726259,4,Monitor(wav,${CALLPATH}/${CALLTIME}-${CALLFILENAME},m)
;exten => 7726259,5,DIAL(SIP/7726259,15,t)
;exten => 7726259,6,Macro(exten-vm,[EMAIL PROTECTED],7726259)
exten => 9873022,1,Macro(exten-vm,[EMAIL PROTECTED],9873022)
"extentions_custom.conf"
[ext-local-custom]
;test to see if this stays
exten => 7726259,1,SetVar(CALLFILENAME=${CALLERIDNUM})
exten => 7726259,2,SetVar(CALLTIME=${DATETIME})
exten => 7726259,3,SetVar(CALLPATH=/var/www/html/monitor/7726259)
exten => 7726259,4,Monitor(wav,${CALLPATH}/${CALLTIME}-${CALLFILENAME},m)
exten => 7726259,5,DIAL(SIP/7726259,15,t)
exten => 7726259,6,Macro(exten-vm,[EMAIL PROTECTED],7726259)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Tuesday, April 12, 2005 7:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] overwriting config file problem
extensions_additional.conf and extensions.conf are for
AMP only and should not be changed.
extensions_custom.conf is for user mods. There are
some default modes in there already. It is posible to
do almost anything from extensions_custom.conf because
as you noticed all of the AMP contexts have an include
like "include => ext-local-custom" to link them to
extensions_custom.conf
--- Robert Webb <[EMAIL PROTECTED]> wrote:
>
> On Tue, 12 Apr 2005 14:05:06 -0500
> "mr. barker" <[EMAIL PROTECTED]> wrote:
> > I am using [EMAIL PROTECTED]
> >
> >
> >
> > When I manually add anything to the
> >extensions_additional.conf file it gets
> > rewritten when I add an extension using the web
> >interface
> >
> > I am trying to include the monitor function .. I
> got
> >that working however it
> > gets deleted when I add something using the web
> >interface
> >
> >
> >
> > I see that it can "include => ext-local-custom"
> is this
> >the file that
> > should be used to add custom scripting ? If so
> where
> >would it be located?
> >
> >
>
> It would be located in the /etc/asterisk directory.
> It is
> not created by default, that I can see, so you will
> need
> to create one and add your cusom config in it. Or
> else
> just modify the extensions.conf file and add it
> there then
> do an include of your custom section.
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[Asterisk-Users] overwriting config file problem
I am using [EMAIL PROTECTED] When I manually add anything to the extensions_additional.conf file it gets rewritten when I add an extension using the web interface I am trying to include the monitor function .. I got that working however it gets deleted when I add something using the web interface I see that it can “include => ext-local-custom” is this the file that should be used to add custom scripting ? If so where would it be located? [ext-local] include => ext-local-custom exten => 7726257,1,Macro(exten-vm,[EMAIL PROTECTED],7726257) exten => 7726258,1,Macro(exten-vm,[EMAIL PROTECTED],7726258) exten => 7726259,1,Macro(exten-vm,[EMAIL PROTECTED],7726259) exten => 9873022,1,Macro(exten-vm,[EMAIL PROTECTED],9873022) exten => 9873023,1,Macro(exten-vm,novm,9873023) Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dumb question ?
Here it is
exten => s,1,answer
exten => s,2,SetCIDName('PMG')
In a lot of config files I see "exten => s,"snip ..
Is "s" just an extension or system variable for all extensions ? or
something else ?
Thanks
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[Asterisk-Users] RE: Monitor with Asterisk@Home
Thank you for the reply.
exten => 1,1,SetVar(CALLFILENAME=${CALLERIDNUM})
exten => 1,2,SetVar(CALLTIME=${DATETIME})
exten => 1,3,SetVar(CALLPATH=/var/calls)
exten => 1,4,Monitor(wav,${CALLPATH}/${CALLTIME}-${CALLFILENAME},m)
? exten => 1,5,DIAL(SIP/something,15,t) - do I need to change SIP/something
exten => 1,6,StopMonitor
? exten => 1,7,Voicemail(u804) - what does u804 stand for or do
exten => 1,8,Hangup
exten => 1,102,StopMonitor
? exten => 1,103,VoiceMail(b804) -
exten => 1,104,Hangup
Would I also change
exten => 1,... to reflect the extention # if I am not using 1 as an
extension ie. Exten=> 7726259 or
Do I put the above in the [ext-local] after each exten or does it get placed
in the
[ext-local]
include => ext-local-custom
exten => 7726257,1,Macro(exten-vm,[EMAIL PROTECTED],7726257)
exten => 7726258,1,Macro(exten-vm,[EMAIL PROTECTED],7726258)
exten => 7726259,1,Macro(exten-vm,[EMAIL PROTECTED],7726259)
exten => 9873022,1,Macro(exten-vm,[EMAIL PROTECTED],9873022)
exten => 9873023,1,Macro(exten-vm,novm,9873023)
or in the
[aa_1]
include => aa_1-custom
exten => 1,1,Goto(ext-local,7726258,1) ;
exten => 2,1,Goto(ext-local,7726259,1) ;
exten => 3,1,Goto(ext-local,7726257,1) ;
exten => fax,1,Goto(ext-fax,in_fax,1) ;
- snip -
Thankyou in return.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Sent: Monday, April 11, 2005 8:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Monitor with [EMAIL PROTECTED]
You have to put the monitor after the person presses their selection.
This is how ours is:
exten => s,1,answer
exten => s,2,SetCIDName('PMG')
exten => s,3,SetVar(company=PMG)
exten => s,4,Wait(1)
exten => s,5,DigitTimeout,5
exten => s,6,ResponseTimeout,40
exten => s,7,Background(/var/lib/asterisk/sounds/greetings/pmg)
exten => s,8,Background(greetings/dial)
exten => 1,1,SetVar(CALLFILENAME=${CALLERIDNUM})
exten => 1,2,SetVar(CALLTIME=${DATETIME})
exten => 1,3,SetVar(CALLPATH=/var/calls)
exten => 1,4,Monitor(wav,${CALLPATH}/${CALLTIME}-${CALLFILENAME},m)
exten => 1,5,DIAL(SIP/something,15,t)
exten => 1,6,StopMonitor
exten => 1,7,Voicemail(u804)
exten => 1,8,Hangup
exten => 1,102,StopMonitor
exten => 1,103,VoiceMail(b804)
exten => 1,104,Hangup
Kyle
mr. barker wrote:
> I am sure that this was answered somewhere but my lack of being able
> to find an answer using google I turn to the pros.
>
>
>
> What would be the easist way to record all conversations using Monitor
> command with the latest [EMAIL PROTECTED] ?
>
> Using a FXO card with SIP extensions
>
>
>
> I have tried adding the following in the extensions_additional.conf
> but I am not getting a file generated in the
> /var/spool/asterisk/monitor directory or anywhere else.
>
> Help would be muchly appreciated.
>
>
>
> Thanks for helping the newbiein return.
>
>
>
> exten => s,7,Monitor(wav,${TIMESTAMP}-${CALLERIDNUM}-${MACRO_EXTEN})
>
>
>
>
>
> [aa_1]
>
> include => aa_1-custom
>
> exten => 1,1,Goto(ext-local,7726258,1) ;
>
> exten => 2,1,Goto(ext-local,7726259,1) ;
>
> exten => 3,1,Goto(ext-local,7726257,1) ;
>
> exten => fax,1,Goto(ext-fax,in_fax,1) ;
>
> exten => h,1,Hangup();
>
> exten => i,1,Playback(invalid) ;
>
> exten => i,2,Goto(s,7);
>
> include => ext-local
>
> include => app-messagecenter
>
> include => app-directory
>
> exten => s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4) ;
>
> exten => s,2,Answer();
>
> exten => s,3,Wait(1) ;
>
> exten => s,4,SetVar(DIR-CONTEXT=default);
>
> exten => s,5,DigitTimeout(3) ; Select
>
> exten => s,6,ResponseTimeout(7) ;
>
> exten => s,7,Monitor(wav,${TIMESTAMP}-${CALLERIDNUM}-${MACRO_EXTEN})
>
> exten => s,8,Background(custom/aa_1) ; Press 1 for Peter Press 2 for
> Paula Press 3 for the Kids
>
>
>
>
>
>
>
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[Asterisk-Users] Monitor with Asterisk@Home
I am sure that this was answered somewhere but my lack of
being able to find an answer using google I turn to the pros.
What would be the easist way to record all conversations
using Monitor command with the latest [EMAIL PROTECTED] ?
Using a FXO card with SIP extensions
I have tried adding the following in the extensions_additional.conf
but I am not getting a file generated in the /var/spool/asterisk/monitor directory or anywhere else.
Help would be muchly
appreciated.
Thanks for helping the
newbiein return.
exten =>
s,7,Monitor(wav,${TIMESTAMP}-${CALLERIDNUM}-${MACRO_EXTEN})
[aa_1]
include => aa_1-custom
exten => 1,1,Goto(ext-local,7726258,1) ;
exten => 2,1,Goto(ext-local,7726259,1) ;
exten => 3,1,Goto(ext-local,7726257,1) ;
exten => fax,1,Goto(ext-fax,in_fax,1) ;
exten => h,1,Hangup() ;
exten => i,1,Playback(invalid) ;
exten => i,2,Goto(s,7) ;
include => ext-local
include => app-messagecenter
include => app-directory
exten => s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4) ;
exten => s,2,Answer() ;
exten => s,3,Wait(1) ;
exten => s,4,SetVar(DIR-CONTEXT=default) ;
exten => s,5,DigitTimeout(3) ; Select
exten => s,6,ResponseTimeout(7) ;
exten =>
s,7,Monitor(wav,${TIMESTAMP}-${CALLERIDNUM}-${MACRO_EXTEN})
exten => s,8,Background(custom/aa_1) ; Press 1 for
Peter Press 2 for Paula Press 3 for the Kids
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RE: [Asterisk-Users] Digtial Receptionist Recorded Greeting LocationProblem
It is in the the compressed file of AMP called upgrade. Instructions are also in the file. http://amp.coalescentsystems.ca/ From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 05, 2005 11:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Digtial Receptionist Recorded Greeting LocationProblem Where did you find the upgrade script? I have not been able to find this. THanks! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mr. barker Sent: Tuesday, April 05, 2005 9:25 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Digtial Receptionist Recorded Greeting LocationProblem I am using the latest AMP with [EMAIL PROTECTED] – I used the AMP upgrade script My problem is this … I am able to record a greeting using *77 and it put the recording into the /var/lib/asterisk/sounds location with the extentsion preceding the recording However it seems that asterisk is looking for the file name to be called Unavail.gsm and to be located in the voicemail directory of the extention. When the extension is called and goes to voicemail the greeting is not played but instead the extension number (using the number(s).gsm) audio. I am wondering how would I be able to have the greeting stored in the extension’s directory for the voicemail greeting. The work around that I am currently doing is just renaming the file and placing it where it is looking for it. Thanks in return. Ctl-alt-del ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digtial Receptionist Recorded Greeting Location Problem
I am using the latest AMP with [EMAIL PROTECTED] – I used the AMP upgrade script My problem is this … I am able to record a greeting using *77 and it put the recording into the /var/lib/asterisk/sounds location with the extentsion preceding the recording However it seems that asterisk is looking for the file name to be called Unavail.gsm and to be located in the voicemail directory of the extention. When the extension is called and goes to voicemail the greeting is not played but instead the extension number (using the number(s).gsm) audio. I am wondering how would I be able to have the greeting stored in the extension’s directory for the voicemail greeting. The work around that I am currently doing is just renaming the file and placing it where it is looking for it. Thanks in return. Ctl-alt-del ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
