Re: [asterisk-users] Configure Voicemail for Large Systems
Would it be possible to see an example on extensions.conf and voicemail.conf to see how to do that? Thanks in advance, Jonathan On Mon, Jun 14, 2010 at 10:18 PM, Zeeshan Zakaria zisha...@gmail.comwrote: You can use realtime architecture. I have a similar setup, voicemails works just fine. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-14 6:08 PM, Jonathan González jonathan@gmail.com wrote: Hi there, I have been taking a look on how to configure voicemail systems with asterisk and I would like to know if there's any way to define mailbox in a dynamic way. I have 100 users and I would like to know if there's any way to avoid the definition of the 100 mailboxes in voicemail.conf and use for example the variables that I am getting from extensions.conf. Thanks in advance, Jonathan -- Personal webpage - www.jonbaraq.eu -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Personal webpage - www.jonbaraq.eu -- _ -- 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] Unable to pickup an extension, trying everything
I have read some info on PICKUPMARK and that I need to set this when a call comes in. But what happens when there are multiple calls coming in ?? How will Pickup(1...@pickupmark) know which channel to pick up ?? In stead of PICKUPMARK (which is a global variable) I would rather like to use a more context-sensitive approach if possible. Jonas. On 06/15/2010 01:34 AM, Philipp von Klitzing wrote: Quickly: Do some reading on PICKUPMARK: You need to set this on the channel that you want to pick up, not the channel that is doing the pickup. Philipp -- _ -- 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] a2billing for residential voip usage
Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ram talk2...@gmail.com wrote: From: ram talk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landy landysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- 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 -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
I send you my a2b config for whole sale use_dnid = YES - this is the main option that you must use You can call this config like so: DeadAGI(a2billing.php|3) I hope this will be help you. [agi-conf3] ; the debug level ; 0=none, 1=low, 2=normal, 3=all debug = 0 ; Asterisk Version Information ; 1_1,1_2,1_4 By Default it will take 1_2 or higher asterisk_version = 1_4 ; Manage the answer on the call answer_call = NO ; Play audio - this will disable all stream file but not the Get Data ; for wholesale ensure that the authentication works and than number_try = 1 play_audio = NO ; play the goodbye message when the user has finished. say_goodbye = NO ; enable the menu to choose the language ; press 1 for English, pulsa 2 para el espaУБol, Pressez 3 pour FranУЇais play_menulanguage = NO ; force the use of a language, if you dont want to use it leave the option empty ; Values : ES, EN, FR, etc... (according to the audio you have installed) force_language = ; Introduction prompt : to specify an additional prompt to play at the beginning of the application intro_prompt = ; Minimum amount of credit to use the application min_credit_2call = 0 ; this is the minimum duration in seconds of a call in order to be billed ; any call with a length less than min_duration_2bill will have a 0 cost ; useful not to charge callers for system errors when a call was answered but it actually didn't connect min_duration_2bill = 0 ; if user doesn't have enough credit to call a destination, prompt him to enter another cardnumber notenoughcredit_cardnumber = NO ; if notenoughcredit_cardnumber = YES then assign the CallerID to the new cardnumber notenoughcredit_assign_newcardnumber_cid = NO ; if YES it will use the DNID and try to dial out, without asking for the phonenumber to call ; value : YES, NO use_dnid = YES ; list the dnid on which you want to avoid the use of the previous option use_dnid no_auth_dnid = 2400,2300 ; number of times the user can dial different number number_try = 1 ; this will force to select a specific call plan by the Rate Engine force_callplan_id = ; Play the balance to the user after the authentication (values : yes - no) say_balance_after_auth = NO ; Play the balance to the user after the call (values : yes - no) say_balance_after_call = NO ; Play the initial cost of the route (values : yes - no) say_rateinitial = NO ; Play the amount of time that the user can call (values : yes - no) say_timetocall = NO ; enable the setup of the callerID number before the outbound is made, by default the user callerID value will be use auto_setcallerid = NO ; If auto_setcallerid is enabled, the value of force_callerid will be set as CallerID force_callerid = ; If force_callerid is not set, then the following option ensures that CID is set to one of the card's configured caller IDs or blank if none available. ; NO - disable this feature, caller ID can be anything. ; CID - Caller ID must be one of the customers caller IDs ; DID - Caller ID must be one of the customers DID nos. ; BOTH - Caller ID must be one of the above two items. cid_sanitize = NO ; enable the callerid authentication ; if this option is active the CC system will check the CID of caller cid_enable = NO ; if the CID does not exist, then the caller will be prompt to enter his cardnumber cid_askpincode_ifnot_callerid = NO ; if the callerID authentication is enable and the authentication fails then the user will be prompt to enter his cardnumber ; this option will bound the cardnumber entered to the current callerID so that next call will be directly authenticate cid_auto_assign_card_to_cid = NO ; if the callerID is captured on a2billing, this option will create automatically a new card and add the callerID to it cid_auto_create_card = NO ; set the length of the card that will be auto create (ie, 10) cid_auto_create_card_len = 10 ; If cid_auto_create_card has been set to YES, the following options will define with which configuration we will create the card ; ; billing type of the new card ; ( value : POSTPAY or PREPAY) cid_auto_create_card_typepaid = POSTPAY ; amount of credit of the new card cid_auto_create_card_credit = 0 ; if postpay, define the credit limit for the card cid_auto_create_card_credit_limit = 1000 ; the tariffgroup to use for the new card (this is the ID that you can find on the admin web interface) cid_auto_create_card_tariffgroup = 6 ; to check callerID over the cardnumber authentication (to guard against spoofing) callerid_authentication_over_cardnumber = NO ; enable the option to call sip/iax friend for free (values : YES - NO) sip_iax_friends = no ; if SIP_IAX_FRIENDS is active, you can define a prefix for the dialed digits to call a pstn number ; values : number sip_iax_pstn_direct_call_prefix = 555 ; this will enable a prompt to enter your destination number. ; if number start by sip_iax_pstn_direct_call_prefix we do directly a sip iax call, if not we do a normal call
Re: [asterisk-users] Skype for SIP
On 15/6/10 06:22, Randy R wrote: By the way, I am currently testing this product from Skype. I would like to be able to receive calls ona Skype name on our pbx. 1) It works beautifully and you don't have to do anything in particular. 2) It's disproportionally expensive which is why I want Skype for Asterisk to work. SfS costs $5 per month per channel just to test the beta! I find that insane, but I wanted to test it. In October, they will begin charging for Skype Manager (required for SfS) and a per seat charge for that. SfA also requires Skype Manager, and only works with users that were created with it. (At Skypes insistance afaict). The only architectures supported by SfA at the moment are x86 and x86-64. Also afaik, video still doesn't work with it. -- _ -- 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] can't seem to register, status unmonitored
Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations sip:2...@172.26.48.208 sip%3a2...@172.26.48.208;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it shows the user and the ip but status is unmonitored. debian-te410*CLI sip show peers Name/username HostDyn Nat ACL Port Status wlg-gateway202.7.4.40 5060 Unmonitored 2002/2002 (Unspecified)D N 0Unmonitored 2001/2001 172.26.48.113D N 5062 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 1 offline] 113 is my ip. This may be the reason that when i do 'sip show registry' no value is displayed even though i get message of successful registration on my sofphone. debian-te410*CLI sip show registry HostUsername Refresh State Reg.Time Please help, what may be the problem here, should the status be different? I want to make a call from server to the 2001 user through a php file, how can I do so?? Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- 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] Skype for SIP
On Tue, Jun 15, 2010 at 9:46 AM, Thomas Kenyon dig...@sanguinarius.co.uk wrote: On 15/6/10 06:22, Randy R wrote: In October, they will begin charging for Skype Manager (required for SfS) and a per seat charge for that. SfA also requires Skype Manager, and only works with users that were created with it. (At Skypes insistance afaict). Yes, correct. And they are charging per name starting this fall. SO if the names die when you don't pay for Skype Manager, screw it and Skype which I personally never use, that will become expensive for little benefit. We'll teach people to use SIP clients which are free or fixed cost. The only architectures supported by SfA at the moment are x86 and x86-64. ok. Also afaik, video still doesn't work with it. Don't care about that. -- _ -- 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] a2billing for residential voip usage
And also, what a2b version you are use? If you are use 1.7 then all config is in DB, if 1.3(4) all config in a2billing.conf -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Vardan Harutyunyan wrote: I send you my a2b config for whole sale use_dnid = YES - this is the main option that you must use You can call this config like so: DeadAGI(a2billing.php|3) I hope this will be help you. [agi-conf3] ; the debug level ; 0=none, 1=low, 2=normal, 3=all debug = 0 ; Asterisk Version Information ; 1_1,1_2,1_4 By Default it will take 1_2 or higher asterisk_version = 1_4 ; Manage the answer on the call answer_call = NO ; Play audio - this will disable all stream file but not the Get Data ; for wholesale ensure that the authentication works and than number_try = 1 play_audio = NO ; play the goodbye message when the user has finished. say_goodbye = NO ; enable the menu to choose the language ; press 1 for English, pulsa 2 para el espaУБol, Pressez 3 pour FranУЇais play_menulanguage = NO ; force the use of a language, if you dont want to use it leave the option empty ; Values : ES, EN, FR, etc... (according to the audio you have installed) force_language = ; Introduction prompt : to specify an additional prompt to play at the beginning of the application intro_prompt = ; Minimum amount of credit to use the application min_credit_2call = 0 ; this is the minimum duration in seconds of a call in order to be billed ; any call with a length less than min_duration_2bill will have a 0 cost ; useful not to charge callers for system errors when a call was answered but it actually didn't connect min_duration_2bill = 0 ; if user doesn't have enough credit to call a destination, prompt him to enter another cardnumber notenoughcredit_cardnumber = NO ; if notenoughcredit_cardnumber = YES then assign the CallerID to the new cardnumber notenoughcredit_assign_newcardnumber_cid = NO ; if YES it will use the DNID and try to dial out, without asking for the phonenumber to call ; value : YES, NO use_dnid = YES ; list the dnid on which you want to avoid the use of the previous option use_dnid no_auth_dnid = 2400,2300 ; number of times the user can dial different number number_try = 1 ; this will force to select a specific call plan by the Rate Engine force_callplan_id = ; Play the balance to the user after the authentication (values : yes - no) say_balance_after_auth = NO ; Play the balance to the user after the call (values : yes - no) say_balance_after_call = NO ; Play the initial cost of the route (values : yes - no) say_rateinitial = NO ; Play the amount of time that the user can call (values : yes - no) say_timetocall = NO ; enable the setup of the callerID number before the outbound is made, by default the user callerID value will be use auto_setcallerid = NO ; If auto_setcallerid is enabled, the value of force_callerid will be set as CallerID force_callerid = ; If force_callerid is not set, then the following option ensures that CID is set to one of the card's configured caller IDs or blank if none available. ; NO - disable this feature, caller ID can be anything. ; CID - Caller ID must be one of the customers caller IDs ; DID - Caller ID must be one of the customers DID nos. ; BOTH - Caller ID must be one of the above two items. cid_sanitize = NO ; enable the callerid authentication ; if this option is active the CC system will check the CID of caller cid_enable = NO ; if the CID does not exist, then the caller will be prompt to enter his cardnumber cid_askpincode_ifnot_callerid = NO ; if the callerID authentication is enable and the authentication fails then the user will be prompt to enter his cardnumber ; this option will bound the cardnumber entered to the current callerID so that next call will be directly authenticate cid_auto_assign_card_to_cid = NO ; if the callerID is captured on a2billing, this option will create automatically a new card and add the callerID to it cid_auto_create_card = NO ; set the length of the card that will be auto create (ie, 10) cid_auto_create_card_len = 10 ; If cid_auto_create_card has been set to YES, the following options will define with which configuration we will create the card ; ; billing type of the new card ; ( value : POSTPAY or PREPAY) cid_auto_create_card_typepaid = POSTPAY ; amount of credit of the new card cid_auto_create_card_credit = 0 ; if postpay, define the credit limit for the card cid_auto_create_card_credit_limit = 1000 ; the tariffgroup to use for the new card (this is the ID that you can find on the admin web interface) cid_auto_create_card_tariffgroup = 6 ; to check callerID over the cardnumber authentication (to guard against spoofing)
[asterisk-users] Corba interface
Hi there Has anyone know how to configure asterisk to be able to query Corba interface directly from the dialplan Sam -- _ -- 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] Skype for Asterisk - what processors/platforms does it run on?
Hi! I understand that SfA is a binary module? There are processors it will not work on, correct? Are there limits as to operating system or distros? Requires kernel 2.6, does not work on 2.4 - as I had to find out the hard way (this is not documented - so embedded people: be aware!). Philipp -- _ -- 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] Configure Voicemail for Large Systems
Its all there on voip-info.org. search for realtime architecture. However you'll need some GUI to interact with the MySQL database. When using realtime, you don't deal with voicemail.conf, and will need to write a dialplan which could interact with the database. It'll need some learning though, but its not very hard for a simple setup and voip-info.org has all the info for it, that's where I learned it from. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-15 2:52 AM, Jonathan González jonathan@gmail.com wrote: Would it be possible to see an example on extensions.conf and voicemail.conf to see how to do that? Thanks in advance, Jonathan On Mon, Jun 14, 2010 at 10:18 PM, Zeeshan Zakaria zisha...@gmail.com wrote: You can use real... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't seem to register, status unmonitored
1. Do 'qualify=yes' in sip.conf for the extenstions for which you want to see the status. 2. 'sip show registry' doesn't show anything for the extensions registering on your server, it shows your server registering on another server, i.e. when when setting up a trunk. 3. Using php to make a call, you need to dedicate some time (probably a week) for learning AGI using phpagi. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-15 4:11 AM, nikhil singhania niksingha...@gmail.com wrote: Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations sip:2...@172.26.48.208 sip%3a2...@172.26.48.208;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it shows the user and the ip but status is unmonitored. debian-te410*CLI sip show peers Name/username HostDyn Nat ACL Port Status wlg-gateway202.7.4.40 5060 Unmonitored 2002/2002 (Unspecified)D N 0Unmonitored 2001/2001 172.26.48.113D N 5062 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 1 offline] 113 is my ip. This may be the reason that when i do 'sip show registry' no value is displayed even though i get message of successful registration on my sofphone. debian-te410*CLI sip show registry HostUsername Refresh State Reg.Time Please help, what may be the problem here, should the status be different? I want to make a call from server to the 2001 user through a php file, how can I do so?? Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk reject SIP INTITE from different source ports
Hi, On some SIP interconnects with devices like Cisco, Dialogic we get SIP invite from different source port every time and asterisk rejects that INVITE. Does anyone knows solution for this? --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151 Email: deepika.nijha...@oxygen8.com Skype: deepika-nijhawan W: http://www.oxygen8.com/ www.oxygen8.com This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the intended recipient/s. If you are not the intended recipient/s please note that any distribution, copying or use of this communication or the information in it is strictly prohibited. If you have received this communication in error please notify us by email or by telephone (08082060808) and then delete the email and any copies of it. This communication is from Oxygen8 Communications UK Ltd - Company Number 03383285. Registered Address; 12th Floor, Lyndon House, 58-62 Hagley Road, Birmingham, B16 8PE. VAT Registration Number: 792 4494 89 -- _ -- 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] can't seem to register, status unmonitored
Hi Zeeshan, Thanx for ur reply!! The reason for this question was that i am actually doing the 3rd part, which you said will take me 1 week to learn. I have modified a file inbound.php which uses function of phpagi.phpexec_dial. But since i am not able to get the call on softphone. Here is part of code: $agi = new AGI(); $agi-answer(); $agi-exec_dial(SIP,2001); when i execute the php file on the command line of server, nothing happens in my softphone. Since it's registered as i told you then when the file is executed at server, my phone is supposed to ring , but its not ringing. Where I am going wrong?? Message: 19 Date: Tue, 15 Jun 2010 07:01:43 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] can't seem to register, status unmonitored To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: aanlktil6aaf21hcg4jpf7sv9yzpja7w-yo8st6ppf...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 1. Do 'qualify=yes' in sip.conf for the extenstions for which you want to see the status. 2. 'sip show registry' doesn't show anything for the extensions registering on your server, it shows your server registering on another server, i.e. when when setting up a trunk. 3. Using php to make a call, you need to dedicate some time (probably a week) for learning AGI using phpagi. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-15 4:11 AM, nikhil singhania niksingha...@gmail.com wrote: Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations sip:2...@172.26.48.208 sip%3a2...@172.26.48.208 sip%3a2...@172.26.48.208 sip%253a2...@172.26.48.208;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it shows the user and the ip but status is unmonitored. debian-te410*CLI sip show peers Name/username HostDyn Nat ACL Port Status wlg-gateway202.7.4.40 5060 Unmonitored 2002/2002 (Unspecified)D N 0Unmonitored 2001/2001 172.26.48.113D N 5062 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 1 offline] 113 is my ip. This may be the reason that when i do 'sip show registry' no value is displayed even though i get message of successful registration on my sofphone. debian-te410*CLI sip show registry HostUsername Refresh State Reg.Time Please help, what may be the problem here, should the status be different? I want to make a call from server to the 2001 user through a php file, how can I do so?? Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- 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 -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100615/79ebe9fb/attachment.htm -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2010 - October 26-28 Washington, DC Register Now: http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 71, Issue 33 ** -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- 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] a2billing for residential voip usage
Hi, Maybe you can just use a reporting tool that will look at the CDR and tell you who's using the phone the most. Some of them will use a DB to store the CDR. If you want, you can even use Excel to look at the csv file created by default and make your own report. http://www.voip-info.org/wiki/view/Asterisk+billing http://www.voip-info.org/wiki/view/Asterisk+GUI (in Billing Call Detail Reporting) http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Jimmy -Original Message- From: landysacco...@yahoo.com Sent: Tue, 15 Jun 2010 00:11:51 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] a2billing for residential voip usage Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ram talk2...@gmail.com wrote: From: ram talk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landy landysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- 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 -Inline Attachment Follows- -- _ -- 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 Share photos screenshots in seconds... TRY FREE IM TOOLPACK at http://www.imtoolpack.com/default.aspx?rc=if1 Works in all emails, instant messengers, blogs, forums and social networks. -- _ -- 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] Small PC to build and run Asterisk
Danny Nicholas wrote: Also cheaper to replace flash card than hard drive. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Monday, June 14, 2010 4:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Small PC to build and run Asterisk Why no flash? * Small pre-built PC (not buying board, case, all parts separately) * Low power consumption * No fan or very small fan * Hard drive (not flash memory) An ssd uses less power, so generates less warmth, hence less need for fan in the drive area. Also less noise.. I like this one, or its smaller brother: http://www.fit-pc.com/web/fit-pc2/fit-pc2i-specifications/ But a flash card needs replacing more often than a hard drive. It's just not designed for the same sort of lifecycle of writes that a hard drive is. Sure, the number is always increasing as they increase the capacity, but it WILL NOT LAST. Dependent on the type of filesystem access you need, SSD could be a great choice. But if you're heavy on logging and writing small data bits here and there (which isn't always something you can control if you don't write all the software), then a hard drive is just going to be the better choice to hold up for a long period of time. N. -- _ -- 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] a2billing for residential voip usage
look manual, but in any case the a2billing.conf is in /etc/asterisk/ on can say, where you have place your asterisk configuration files -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Jimmy Godbout wrote: Hi, Maybe you can just use a reporting tool that will look at the CDR and tell you who's using the phone the most. Some of them will use a DB to store the CDR. If you want, you can even use Excel to look at the csv file created by default and make your own report. http://www.voip-info.org/wiki/view/Asterisk+billing http://www.voip-info.org/wiki/view/Asterisk+GUI (in Billing Call Detail Reporting) http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Jimmy -Original Message- From: landysacco...@yahoo.com Sent: Tue, 15 Jun 2010 00:11:51 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] a2billing for residential voip usage Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ramtalk2...@gmail.com wrote: From: ramtalk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landylandysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- 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 -Inline Attachment Follows- -- _ -- 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 Share photos screenshots in seconds... TRY FREE IM TOOLPACK at http://www.imtoolpack.com/default.aspx?rc=if1 Works in all emails, instant messengers, blogs, forums and social networks. -- _ -- 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] Unable to pickup an extension, trying everything
What happens in my extensions.conf is that an incoming call makes a
group of SIPaccounts ring :
[sub-routing]
snip
exten = s,n(group),NoOp()
exten = s,n,Macro(GetGroupDetails,${ganaarID})
exten = s,n,GoToIf($[${sequencenr}==1]?callit)
exten = s,n,Answer()
exten = s,n(callit),Dial(${SIPaccounts},${timeout})
exten = s,n,GoToIf($[${DIALSTATUS}!=ANSWER]?nextstep:hangup)
snip
sub-routing is a sort of loop that checks al the steps of routing. A
step can be to dial a SIPaccount or a group of SIPaccounts.
So if this SIPaccount or this group of SIPaccounts are ringing, how can
I pick them up from another extension/IPphone ???
Jonas.
On 06/14/2010 08:00 PM, Peder wrote:
sip.conf and extensions.conf would be helpful as well as knowing what
version you are running. Based on what you went, I would say you have
a config error, but I can't tell where without seeing the config.
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Re: [asterisk-users] Unable to pickup an extension, trying everything
Hi! But what happens when there are multiple calls coming in ?? How will Pickup(1...@pickupmark) know which channel to pick up ?? In stead of PICKUPMARK (which is a global variable) I would rather like to use a more context-sensitive approach if possible. Do not make PICKUPMARK a global variable - use it as a normal channel variable instead (and prefix it with _ (or even __) when you see fit to provide channel variable inheritance). Philipp -- _ -- 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] Unable to pickup an extension, trying everything
Philipp,
thank you for your willingness to help me.
In a previous mail I gave a part of my dialplan: an incoming call rings
a group of extensions/SIPaccounts :
[sub-routing]
snip
exten = s,n(group),NoOp()
exten = s,n,Macro(GetGroupDetails,${ganaarID})
exten = s,n,GoToIf($[${sequencenr}==1]?callit)
exten = s,n,Answer()
exten = s,n(callit),Dial(${SIPaccounts},${timeout})
exten = s,n,GoToIf($[${DIALSTATUS}!=ANSWER]?nextstep:hangup)
snip
So the following happens :
exten = s,n,Dial(SIP/testcorp1SIP/testcorp2) ; dial multiple ext
or
exten = s,n,Dial(SIP/testcorp1) ; dial one ext
I want to pick up the calling extension, in this case extension 10
(testcorp1) or 20 (testcorp2), or extension 10 (testcorp1).
How to do this ??
To proceed with your answer on PICKUPMARK, where do I put this ???
Remark : internal calls no problem (like calling extension 10 from
extension 20, and pick up extension 10 from extension 30)
But calls coming from external... impossible !
Jonas.
On 06/15/2010 02:38 PM, Philipp von Klitzing wrote:
Hi!
Do not make PICKUPMARK a global variable - use it as a normal channel
variable instead (and prefix it with _ (or even __) when you see fit to
provide channel variable inheritance).
Philipp
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[asterisk-users] Cutting the CallerID(RDNIS)
We are having an issue with the RDNIS coming through with a leading 1 on some calls. I have been trying to find a way to remove the leading number only if it starts with a 1 and have yet to find a solid solution. If anyone else has any idea as how to do this I would greatly appreciate it! Thanks. -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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] can't seem to register, status unmonitored
On Tue, 15 Jun 2010, nikhil singhania wrote: I have modified a file inbound.php which uses function of phpagi.phpexec_dial. But since i am not able to get the call on softphone. when i execute the php file on the command line of server, nothing happens in my softphone. Since it's registered as i told you then when the file is executed at server, my phone is supposed to ring , but its not ringing. Where I am going wrong?? You cannot execute an AGI that executes dial() from the command line. You can debug an AGI from the command line by feeding the proper responses into STDIN, but it cannot interact with the running instance of Asterisk. -- 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 -- 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] Corba interface
Hi there Does anyone know how to configure asterisk to be able to query Corba interface directly from the dialplan Sam -- _ -- 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 reject SIP INTITE from different sourceports
We have two models of Cisco phone that do this, but Asterisk handles it fine. I don't recall doing anything special to make it work. We're using Asterisk v1.4.17. What is the error message on Asterisk? -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Deepika Nijhawan Sent: Tue 6/15/2010 4:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk reject SIP INTITE from different sourceports Hi, On some SIP interconnects with devices like Cisco, Dialogic we get SIP invite from different source port every time and asterisk rejects that INVITE. Does anyone knows solution for this? --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151 Email: deepika.nijha...@oxygen8.com Skype: deepika-nijhawan W: http://www.oxygen8.com/ www.oxygen8.com This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the intended recipient/s. If you are not the intended recipient/s please note that any distribution, copying or use of this communication or the information in it is strictly prohibited. If you have received this communication in error please notify us by email or by telephone (08082060808) and then delete the email and any copies of it. This communication is from Oxygen8 Communications UK Ltd - Company Number 03383285. Registered Address; 12th Floor, Lyndon House, 58-62 Hagley Road, Birmingham, B16 8PE. VAT Registration Number: 792 4494 89 winmail.dat-- _ -- 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] Cutting the CallerID(RDNIS)
Hi,
I do something similar but for outbound calls. But same should work for
inbound calls.
First catch the calls with a leading 1, then strip 1 and send them to your
desired context or extension. I use AEL, but in regular config it should be
something like:
[incoming-calls]
; catch numbers with leading 1 here
exten = _1X.,1,Goto(${EXTEN:1},1)
; numbers with no leading 1 go here
exten = _X.,1,Noop( - - - - - Incoming call - - - - -)
Zeeshan A Zakaria
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On 2010-06-15 9:27 AM, Chris Ramirez crami...@tele-onecom.com wrote:
We are having an issue with the RDNIS coming through with a leading 1 on
some calls. I have been trying to find a way to remove the leading number
only if it starts with a 1 and have yet to find a solid solution. If anyone
else has any idea as how to do this I would greatly appreciate it! Thanks.
--
*Chris Ramirez*
TELE-ONE COMMUNICATIONS, INC.
crami...@tele-onecom.com
903-531-0777
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[asterisk-users] Asterisk reject SIP INTITE from different
It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' -- _ -- 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 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
Hi, We are using Asterisk 1.6.2 and it is continually failing to resolve Verizon SRV and sending following message, WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' DNS settings on OS level is working fine. Can anyone have an idea about it? Regards, Faisal Hanif -- _ -- 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] Dual Atom mobo - call capacity
On 11/06/10 01:19, Michelle Dupuis wrote: I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? MD Any of the Atom CPU systems will /easily/ handle 25 concurrent calls (and with a 25 extension system, 25 concurrent calls is very unlikely). I use the single core Atom 230 CPUs for systems of this size. Something to bear in mind is how the system will be used, max concurrent calls isn't really that great a performance factor, call arrival rates are more relevant, the CPU time is spent setting up and tearing down calls. Simply having calls in progress with no transcoding uses a tiny amount of CPU in comparison to the work involved setting up and routing a new call. cheers, Paul. -- _ -- 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] HDLC Bad FCS (8) on Primary D-channel
2010/6/11 Karsten Wemheuer k...@gmx.de Hi Olivier, Am Freitag, den 11.06.2010, 14:27 +0200 schrieb Olivier: 2010/6/11 Karsten Wemheuer k...@gmx.de Hi, Am Freitag, den 11.06.2010, 11:54 +0100 schrieb Gareth Blades: Olivier wrote: Hello, I've got a running system in which logs are full of messages such as: [Jun 10 07:24:14] NOTICE[2414] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 The strange thing is those messages are coming from a single span. My setup is : Asterisk 1.6.1.18 Junghanns OctoBRI with wcb4xxp driver libpri 1.4.10.2 dahdi 2.3.0 3 BRI lines in PtMP mode What does this PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 roughly mean ? Why could it happen on a single port and not on the others ? Regards Basically it means that one of the messages it received on the PRI D channel failed the checksum. I take it that in your span command you have 'crc4' or similar specified as an option for all of your spans? If thats the case its probably a faulty port on the card, cable, or a card in the local telephone exchange, AFAIK CRC4 is for PRI only. The setup of Olivier is BRI in PTmP mode. Many providers drive Layer 1 down in case of inactivity. Maybe the driver has a problem with such lines. Would that explain why only a single port is hit ? No, except if this port is configured differently from the others (on the provider site)... This port is the 2nd and the dialing pattern is DAHDI/g1 which means start with channel 1 on span 1, then channel 2 on span 1, then channel 4 on span 2, . Ok, but dialing is Layer 3. You are observing layer 2 errors on the D-Channel (LAPD-protocol). What I observed is that provider sends incoming calls alternatively to each span : if an inbound call comes through span 2 (channel 4 or 5), then the provider would send the next one to span 3 (channel 7 or 8) if available, etc ... To my experience a provider do not send incoming calls to different ports on PTmP lines. Each line gets his own numbers, there is no overflow (at least in germany). But again: Your original problem are layer 2 errors. Reasons could be: - line broken - port broken - driver do not handle layer 1 down in case of inactivity I'll try to swap cables and see if messages are moving from span 2 to another span. Good idea. Tough I swaped cables between ports 2 and 3, I still can read messages such as : chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 My thoughts are this happens because port 2 is somehow broken. So I changed my config to use port 1, 3 and 4 instead of 1,2 and 3, and see if things are improving. I'll report here what I'll find. Have a nice weekend. Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
On Tue, Jun 15, 2010 at 07:50:36PM +0500, Faisal Hanif wrote: Hi, We are using Asterisk 1.6.2 and it is continually failing to resolve Verizon SRV and sending following message, WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' DNS settings on OS level is working fine. Can anyone have an idea about it? I think asterisk only does UDP DNS queries. The response here is too long. $ dig _sip._udp.whsvoip.globalipcom.com SRV ;; Warning: Message parser reports malformed message packet. ;; Truncated, retrying in TCP mode. ; DiG 9.3.4 _sip._udp.whsvoip.globalipcom.com SRV ;; global options: printcmd ;; Got answer: ;; -HEADER- opcode: QUERY, status: NOERROR, id: 63565 ;; flags: qr rd ra; QUERY: 1, ANSWER: 8, AUTHORITY: 0, ADDITIONAL: 0 ;; QUESTION SECTION: ;_sip._udp.whsvoip.globalipcom.com. IN SRV ;; ANSWER SECTION: _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy20.nl.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy00.de.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy00.nl.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy00.uk.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy10.de.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy10.nl.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy10.uk.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy20.de.whsvoip.globalipcom.com. ;; Query time: 3 msec ;; SERVER: 66.92.213.114#53(66.92.213.114) ;; WHEN: Tue Jun 15 11:28:09 2010 ;; MSG SIZE rcvd: 515 -- Barry -- _ -- 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] Priority between calls in different queues
On 6/10/10, Hapworth Slim hpvgtfjdjbpkt...@gmail.com wrote: I'm trying to figure this out. I have agents answer calls from two different queues. We have things set up so that these agents only see one call at a time. Let's say an agent picks up a call while there are calls waiting in both queues. Clearly the head of one of the queues will now start ringing through to the other agents. But which one? Is that something that can be configured, perhaps by saying one queue has priority, or the older call has priority, or something different? Or is it something non-deterministic? I haven't gotten any answers to my questions. Is it because no one has read them or because no one else can figure it out either? I tried looking at the documentation I could locate on the internet, but I couldn't find anything that addressed this issue. The closest thing I could find was something about setting the queue weights, but it didn't say what happens when the weights are equal or not set. -- _ -- 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 reject SIP INTITE from different source ports
It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' -- _ -- 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] Unable to pickup an extension, tryi
Hi! How to do this ?? To proceed with your answer on PICKUPMARK, where do I put this ??? Look at the example for Asterisk 1.4 on this page: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup Philipp -- _ -- 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 reject SIP INTITE from different source ports
Deepika Nijhawan wrote: It just gives no matching peer error and doesn’t pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' So the question is why didnt it match anything. If the phones are registering then they should reregister before choosing a different port. Are they going through a firewall by any chance? -- _ -- 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 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
Hi, I am also wonder that same SRV record is working fine on one machine but not on 2nd while both have same asterisk version. It may be some missing OS utilities which asterisk using to resolve SRV? Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller Sent: Tuesday, June 15, 2010 8:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' On Tue, Jun 15, 2010 at 07:50:36PM +0500, Faisal Hanif wrote: Hi, We are using Asterisk 1.6.2 and it is continually failing to resolve Verizon SRV and sending following message, WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' DNS settings on OS level is working fine. Can anyone have an idea about it? I think asterisk only does UDP DNS queries. The response here is too long. $ dig _sip._udp.whsvoip.globalipcom.com SRV ;; Warning: Message parser reports malformed message packet. ;; Truncated, retrying in TCP mode. ; DiG 9.3.4 _sip._udp.whsvoip.globalipcom.com SRV ;; global options: printcmd ;; Got answer: ;; -HEADER- opcode: QUERY, status: NOERROR, id: 63565 ;; flags: qr rd ra; QUERY: 1, ANSWER: 8, AUTHORITY: 0, ADDITIONAL: 0 ;; QUESTION SECTION: ;_sip._udp.whsvoip.globalipcom.com. IN SRV ;; ANSWER SECTION: _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy20.nl.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy00.de.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy00.nl.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy00.uk.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy10.de.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy10.nl.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy10.uk.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy20.de.whsvoip.globalipcom.com. ;; Query time: 3 msec ;; SERVER: 66.92.213.114#53(66.92.213.114) ;; WHEN: Tue Jun 15 11:28:09 2010 ;; MSG SIZE rcvd: 515 -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
I copied the config to the a2billing.conf in /etc/asterisk folder. I'm still not able to place any calls yet. Looks like I have to read more on how to configure trunks and providers whick got me confused. I'll learn though. --- On Tue, 6/15/10, Vardan Harutyunyan hvarda...@gmail.com wrote: From: Vardan Harutyunyan hvarda...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 8:03 AM look manual, but in any case the a2billing.conf is in /etc/asterisk/ on can say, where you have place your asterisk configuration files -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Jimmy Godbout wrote: Hi, Maybe you can just use a reporting tool that will look at the CDR and tell you who's using the phone the most. Some of them will use a DB to store the CDR. If you want, you can even use Excel to look at the csv file created by default and make your own report. http://www.voip-info.org/wiki/view/Asterisk+billing http://www.voip-info.org/wiki/view/Asterisk+GUI (in Billing Call Detail Reporting) http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Jimmy -Original Message- From: landysacco...@yahoo.com Sent: Tue, 15 Jun 2010 00:11:51 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] a2billing for residential voip usage Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ramtalk2...@gmail.com wrote: From: ramtalk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landylandysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- 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 -Inline Attachment Follows- -- _ -- 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 Share photos screenshots in seconds... TRY FREE IM TOOLPACK at http://www.imtoolpack.com/default.aspx?rc=if1 Works in all emails, instant messengers, blogs, forums and social networks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Hylafax
Hey Guys I have hylafax working about 95% The problem is I have a DID for fax 0742244224 When I receive a fax I see in the log file n 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [0742244...@from-sip-external:1] NoOp(SIP/5060-0a2f7308, Received incoming SIP connection from unknown peer to 0742244224) in new stack [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [0742244...@from-sip-external:2] Set(SIP/5060-0a2f7308, DID=0742244224) in new stack [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [0742244...@from-sip-external:3] Goto(SIP/5060-0a2f7308, s,1) in new stack [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Goto (from-sip-external,s,1) [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [...@from-sip-external:1] GotoIf(SIP/5060-0a2f7308, 1?from-trunk,0742244224,1) in new stack [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Goto (from-trunk,0742244224,1) [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [0742244...@from-trunk:1] Set(SIP/5060-0a2f7308, __FROM_DID=0742244224) in new stack [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [0742244...@from-trunk:2] Gosub(SIP/5060-0a2f7308, app-blacklist-check,s,1) in new stack [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [...@app-blacklist-check:1] GotoIf(SIP/5060-0a2f7308, 0?blacklisted) in new stack [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [...@app-blacklist-check:2] Return(SIP/5060-0a2f7308, ) in new stack [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [0742244...@from-trunk:3] ExecIf(SIP/5060-0a2f7308, 1 ?Set(CALLERID(name)=0282086500)) in new stack [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [0742244...@from-trunk:4] Set(SIP/5060-0a2f7308, FAX_RX=4111) in new stack [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [0742244...@from-trunk:5] Set(SIP/5060-0a2f7308, fax_rx_email=s...@smellyblackdog.com.au) in new stack [Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing [0742244...@from-trunk:6] Answer(SIP/5060-0a2f7308, ) in new stack [Jun 16 02:44:20] VERBOSE[3679] logger.c: -- Executing [0742244...@from-trunk:7] PlayTones(SIP/5060-0a2f7308, ring) in new stack [Jun 16 02:44:20] VERBOSE[3679] logger.c: -- Executing [0742244...@from-trunk:8] NVFaxDetect(SIP/5060-0a2f7308, 0|t) in new stack [Jun 16 02:44:20] DEBUG[3679] app_nv_faxdetect-1.0.6_1.4.c: Preparing detect of fax (waitdur=4ms, sildur=1000ms, mindur=100ms, maxdur=-1ms) [Jun 16 02:44:24] DEBUG[3679] app_nv_faxdetect-1.0.6_1.4.c: Fax detected on SIP/5060-0a2f7308 [Jun 16 02:44:24] NOTICE[3679] app_nv_faxdetect-1.0.6_1.4.c: Redirecting SIP/5060-0a2f7308 to fax extension [Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing [...@from-trunk:1] Goto(SIP/5060-0a2f7308, ext-fax,in_fax,1) in new stack [Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Goto (ext-fax,in_fax,1) [Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing [in_...@ext-fax:1] StopPlayTones(SIP/5060-0a2f7308, ) in new stack [Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing [in_...@ext-fax:2] GotoIf(SIP/5060-0a2f7308, 0?3:analog_fax,1) in new stack [Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Goto (ext-fax,analog_fax,1) [Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing [analog_...@ext-fax:1] GotoIf(SIP/5060-0a2f7308, 0?4:2) in new stack [Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Goto (ext-fax,analog_fax,2) [Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing [analog_...@ext-fax:2] Set(SIP/5060-0a2f7308, DIAL=IAX2/4111) in new stack [Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing [analog_...@ext-fax:3] Dial(SIP/5060-0a2f7308, IAX2/4111/0282086500,20,d) in new stack My FaxDispatch config is #!/bin/sh ## ## FaxDispatch ## (see `man faxrcvd` for moreyyy # The numbers before the paren correspond to asterisk extensions in # extensions.conf case $CALLID4 in # customer DID routing: 0742242442) sendto=...@smellyblackdog.com.au; FILETYPE=pdf;; # everything else goes to default case: *) sendto=...@smellyblackdog.com.au; FILETYPE=pdf;; esac The problem is that it ignores the called number in the did and drops through to the default I have also done the relevant mod to the /etc/asterisk/extensions.conf file as well Any Ideas?? -- _ -- 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 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
On Tue, Jun 15, 2010 at 09:35:37PM +0500, Faisal Hanif wrote: Hi, I am also wonder that same SRV record is working fine on one machine but not on 2nd while both have same asterisk version. It may be some missing OS utilities which asterisk using to resolve SRV? Could be. To test, does replacing whsvoip.globalipcom.com with, say, whs-proxy00.de.whsvoip.globalipcom.com (or whatever is reachable) make it work? What is different about the two machines you've tried? -- Barry -- _ -- 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] a2billing for residential voip usage
You need to copy or soft link a2billing.conf to /etc/ folder as by default latest version search for it in /etc/ Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Tuesday, June 15, 2010 9:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] a2billing for residential voip usage I copied the config to the a2billing.conf in /etc/asterisk folder. I'm still not able to place any calls yet. Looks like I have to read more on how to configure trunks and providers whick got me confused. I'll learn though. --- On Tue, 6/15/10, Vardan Harutyunyan hvarda...@gmail.com wrote: From: Vardan Harutyunyan hvarda...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 8:03 AM look manual, but in any case the a2billing.conf is in /etc/asterisk/ on can say, where you have place your asterisk configuration files -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Jimmy Godbout wrote: Hi, Maybe you can just use a reporting tool that will look at the CDR and tell you who's using the phone the most. Some of them will use a DB to store the CDR. If you want, you can even use Excel to look at the csv file created by default and make your own report. http://www.voip-info.org/wiki/view/Asterisk+billing http://www.voip-info.org/wiki/view/Asterisk+GUI (in Billing Call Detail Reporting) http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Jimmy -Original Message- From: landysacco...@yahoo.com Sent: Tue, 15 Jun 2010 00:11:51 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] a2billing for residential voip usage Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ramtalk2...@gmail.com wrote: From: ramtalk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landylandysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- 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 -Inline Attachment Follows- -- _ -- 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 Share photos screenshots in seconds... TRY FREE IM TOOLPACK at http://www.imtoolpack.com/default.aspx?rc=if1 Works in all emails, instant messengers, blogs, forums and social networks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth
Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports
Try setting insecure=port,invite in sip peer config. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades Sent: Tuesday, June 15, 2010 9:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports Deepika Nijhawan wrote: It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' So the question is why didnt it match anything. If the phones are registering then they should reregister before choosing a different port. Are they going through a firewall by any chance? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Hylafax
On 06/15/2010 12:48 PM, Samantha wrote:
Hey Guys
I have hylafax working about 95%
The problem is I have a DID for fax 0742244224
When I receive a fax I see in the log file
n 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-sip-external:1] NoOp(SIP/5060-0a2f7308, Received
incoming SIP connection from unknown peer to 0742244224) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-sip-external:2] Set(SIP/5060-0a2f7308, DID=0742244224)
in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-sip-external:3] Goto(SIP/5060-0a2f7308, s,1) in new
stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Goto
(from-sip-external,s,1)
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[...@from-sip-external:1] GotoIf(SIP/5060-0a2f7308,
1?from-trunk,0742244224,1) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Goto
(from-trunk,0742244224,1)
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:1] Set(SIP/5060-0a2f7308, __FROM_DID=0742244224)
in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:2] Gosub(SIP/5060-0a2f7308,
app-blacklist-check,s,1) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[...@app-blacklist-check:1] GotoIf(SIP/5060-0a2f7308, 0?blacklisted) in
new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[...@app-blacklist-check:2] Return(SIP/5060-0a2f7308, ) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:3] ExecIf(SIP/5060-0a2f7308, 1
?Set(CALLERID(name)=0282086500)) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:4] Set(SIP/5060-0a2f7308, FAX_RX=4111) in new
stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:5] Set(SIP/5060-0a2f7308,
fax_rx_email=s...@smellyblackdog.com.au) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:6] Answer(SIP/5060-0a2f7308, ) in new stack
[Jun 16 02:44:20] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:7] PlayTones(SIP/5060-0a2f7308, ring) in new
stack
[Jun 16 02:44:20] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:8] NVFaxDetect(SIP/5060-0a2f7308, 0|t) in new
stack
[Jun 16 02:44:20] DEBUG[3679] app_nv_faxdetect-1.0.6_1.4.c: Preparing detect
of fax (waitdur=4ms, sildur=1000ms, mindur=100ms, maxdur=-1ms)
[Jun 16 02:44:24] DEBUG[3679] app_nv_faxdetect-1.0.6_1.4.c: Fax detected on
SIP/5060-0a2f7308
[Jun 16 02:44:24] NOTICE[3679] app_nv_faxdetect-1.0.6_1.4.c: Redirecting
SIP/5060-0a2f7308 to fax extension
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[...@from-trunk:1] Goto(SIP/5060-0a2f7308, ext-fax,in_fax,1) in new
stack
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Goto (ext-fax,in_fax,1)
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[in_...@ext-fax:1] StopPlayTones(SIP/5060-0a2f7308, ) in new stack
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[in_...@ext-fax:2] GotoIf(SIP/5060-0a2f7308, 0?3:analog_fax,1) in new
stack
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Goto (ext-fax,analog_fax,1)
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[analog_...@ext-fax:1] GotoIf(SIP/5060-0a2f7308, 0?4:2) in new stack
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Goto (ext-fax,analog_fax,2)
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[analog_...@ext-fax:2] Set(SIP/5060-0a2f7308, DIAL=IAX2/4111) in new
stack
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[analog_...@ext-fax:3] Dial(SIP/5060-0a2f7308,
IAX2/4111/0282086500,20,d) in new stack
My FaxDispatch config is
#!/bin/sh
##
## FaxDispatch
## (see `man faxrcvd` for moreyyy
# The numbers before the paren correspond to asterisk extensions in
# extensions.conf
case $CALLID4 in
# customer DID routing:
0742242442) sendto=...@smellyblackdog.com.au; FILETYPE=pdf;;
# everything else goes to default case:
*) sendto=...@smellyblackdog.com.au; FILETYPE=pdf;;
esac
The problem is that it ignores the called number in the did and drops
through to the default
I have also done the relevant mod to the /etc/asterisk/extensions.conf file
as well
Any Ideas??
Not sure if the activity above was an instance where it was supposed to
go to 0742242442 but the DID being passed to iaxmodem wasn't 0742242442:
Dial(SIP/5060-0a2f7308, IAX2/4111/0282086500,20,d) in new stack
In this case $CALLID4 is going to be 0282086500. Double check your
extensions.[conf|ael] and make sure the DID that was called is being
passed in your dial command, often like this: Dial(IAX2/4111/${EXTEN})
-Dave
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Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
Till now I am not able to find any difference between both machines. Can you please tell me how I can try to resolve it on OS level using some utility like dig? Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller Sent: Tuesday, June 15, 2010 10:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' On Tue, Jun 15, 2010 at 09:35:37PM +0500, Faisal Hanif wrote: Hi, I am also wonder that same SRV record is working fine on one machine but not on 2nd while both have same asterisk version. It may be some missing OS utilities which asterisk using to resolve SRV? Could be. To test, does replacing whsvoip.globalipcom.com with, say, whs-proxy00.de.whsvoip.globalipcom.com (or whatever is reachable) make it work? What is different about the two machines you've tried? -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to pickup an extension, tryi
Philipp,
I have read this wiki, but I still don't know how to implement it in my
example...
The wiki says :
exten = 1234,1,Set(__PICKUPMARK=1234)
So how do I do this with my dialplan :
exten = s,n,Dial(${SIPaccounts},${timeout})
or translated :
exten = s,n,Dial(SIP/testcorp1SIP/testcorp2)
Can you give me the exact syntax?? Because I really can not filter this
from the wiki.
Looking forward to your answer, thx !
Jonas.
On 06/15/2010 06:09 PM, Philipp von Klitzing wrote:
Hi!
How to do this ??
To proceed with your answer on PICKUPMARK, where do I put this ???
Look at the example for Asterisk 1.4 on this page:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
Philipp
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[asterisk-users] Asterisk hangs up for some calls
Dear list;
I'm trying for forward some calls to an others asterisk using IAX2 protocol.
But My asterisk can forward some calls and for others it hangs up automaticaly.
Before my asterisk was working perfectly, i do not know what is happening!!
When i try directly zoiper with my provider's asterisk it works perfectly.
Here is the output from the cli when i made a call that asterisk hangs up:
Verbosity is at least 3
-- Accepting AUTHENTICATED call from 192.168.1.5:
requested format = unknown,
requested prefs = (ulaw|slin|alaw),
actual format = ulaw,
host prefs = (gsm|ulaw|alaw),
priority = mine
-- Executing [00212675410...@pstn:1] Set(IAX2/#000105-12477,
calleeNumber=011212675410113) in new stack
-- Executing [00212675410...@pstn:2] AGI(IAX2/#000105-12477,
agi://localhost/ManageCalls.agi?when=beforecalleeNumber=011212675410113) in
new stack
-- AGI Script
agi://localhost/ManageCalls.agi?when=beforecalleeNumber=011212675410113
completed, returning 0
-- Executing [00212675410...@pstn:3] Dial(IAX2/#000105-12477,
IAX2/mylo...@pstn/011212675410113||S(348)) in new stack
-- Setting call duration limit to 348 seconds.
-- Called mylo...@pstn/011212675410113
-- Call accepted by 8.17.37.23 (format ulaw)
-- Format for call is ulaw
-- Hungup 'IAX2/pstn-533'
-- No one is available to answer at this time (1:0/0/0)
== Auto fallthrough, channel 'IAX2/#000105-12477' status is 'NOANSWER'
-- Executing [...@pstn:1] DeadAGI(IAX2/#000105-12477,
agi://localhost/ManageCalls.agi?when=after) in new stack
-- AGI Script agi://localhost/ManageCalls.agi?when=after completed,
returning 0
-- Hungup 'IAX2/#000105-12477'
here is my config:
[pstn]
exten=_00X.,1,Set(calleeNumber=011${EXTEN:2})
exten=_00X.,n,AGI(agi://localhost/ManageCalls.agi?when=beforecalleeNumber=${calleeNumber})
exten =_00X.,n,Dial(IAX2/mylo...@pstn/${calleeNumber},,S(${SECONDS-REMAINING}))
exten = h,1,DeadAGI(agi://localhost/ManageCalls.agi?when=after)
Thanks in advance for your help
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Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
What OS are you running on the two systems? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 15, 2010, at 10:40 AM, Faisal Hanif wrote: Till now I am not able to find any difference between both machines. Can you please tell me how I can try to resolve it on OS level using some utility like dig? Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller Sent: Tuesday, June 15, 2010 10:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' On Tue, Jun 15, 2010 at 09:35:37PM +0500, Faisal Hanif wrote: Hi, I am also wonder that same SRV record is working fine on one machine but not on 2nd while both have same asterisk version. It may be some missing OS utilities which asterisk using to resolve SRV? Could be. To test, does replacing whsvoip.globalipcom.com with, say, whs-proxy00.de.whsvoip.globalipcom.com (or whatever is reachable) make it work? What is different about the two machines you've tried? -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 hangs up for some calls
On Tue, 15 Jun 2010, Adil Zaaraoui wrote: I'm trying for forward some calls to an others asterisk using IAX2 protocol. But My asterisk can forward some calls and for others it hangs up automaticaly. 1) What is different about the numbers? Are some international or to countries restricted by your provider? 2) What does your provider say when you tell them a particular destination failed? 3) If you enable IAX debugging, you may get a clue or get output that may be helpful to others. -- 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 -- 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] Re : Asterisk hangs up for some calls
Thanks Steve - Back again :), 1- Asterisk now forwads calls to my cell phone operator, but it does not forward for other international numbers or others operator of my country. There is no restriction of the operator. NOTE: before, MY ASTERISK WORKS PERFECTLY. 2-they said every thing is ok in their system. i tried zoiper, it works perfectly i can make all calls. 3-Here is the output after enbling IAX dugug: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 0ms SCall: 02203 DCall: 0 [192.168.1.5:3048] VERSION : 2 CALLED NUMBER : 00212675410113 CALLING NUMBER : a...@zeaaraoui CALLING NAME: a...@zeaaraoui USERNAME: #000105 FORMAT : 76 CODEC_PREFS : (ulaw|slin|alaw) Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 2ms SCall: 04121 DCall: 02203 [192.168.1.5:3048] AUTHMETHODS : 3 CHALLENGE : 310289772 USERNAME: #000105 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00016ms SCall: 02203 DCall: 04121 [192.168.1.5:3048] USERNAME: #000105 MD5 RESULT : 8D8EA21DDA56CBE0BE83409A1F70FF77 -- Accepting AUTHENTICATED call from 192.168.1.5: requested format = unknown, requested prefs = (ulaw|slin|alaw), actual format = ulaw, host prefs = (gsm|ulaw|alaw), priority = mine -- Executing [00212675410...@pstn:1] Set(IAX2/#000105-4121, calleeNumber=011212675410113) in new stack Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00011ms SCall: 04121 DCall: 02203 [192.168.1.5:3048] FORMAT : 4 -- Executing [00212675410...@pstn:2] AGI(IAX2/#000105-4121, agi://localhost/ManageCalls.agi?when=beforecalleeNumber=011212675410113) in new stack Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00011ms SCall: 02203 DCall: 04121 [192.168.1.5:3048] -- AGI Script agi://localhost/ManageCalls.agi?when=beforecalleeNumber=011212675410113 completed, returning 0 -- Executing [00212675410...@pstn:3] Dial(IAX2/#000105-4121, IAX2/mylo...@pstn/011212675410113||S(348)) in new stack -- Setting call duration limit to 348 seconds. -- Called mylo...@pstn/011212675410113 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 8ms SCall: 02879 DCall: 0 [8.17.37.23:4569] VERSION : 2 CALLED NUMBER : 011212675410113 CODEC_PREFS : (ulaw|gsm|alaw) CALLING NUMBER : #000105 CALLING PRESNTN : 1 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: #000105 LANGUAGE: en USERNAME: myLogin FORMAT : 4 CAPABILITY : 14 ADSICPE : 2 DATE TIME : 2010-06-15 21:15:02 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00016ms SCall: 05978 DCall: 02879 [8.17.37.23:4569] AUTHMETHODS : 3 CHALLENGE : 604023002 USERNAME: myLogin Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00248ms SCall: 02879 DCall: 05978 [8.17.37.23:4569] MD5 RESULT : 3909a6bdcc977397198d45f99a00e5d5 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00236ms SCall: 05978 DCall: 02879 [8.17.37.23:4569] FORMAT : 4 -- Call accepted by 8.17.37.23 (format ulaw) -- Format for call is ulaw Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00236ms SCall: 02879 DCall: 05978 [8.17.37.23:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: HANGUP Timestamp: 00794ms SCall: 05978 DCall: 02879 [8.17.37.23:4569] CAUSE CODE : 16 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00794ms SCall: 02879 DCall: 05978 [8.17.37.23:4569] -- Hungup 'IAX2/pstn-2879' -- No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'IAX2/#000105-4121' status is 'NOANSWER' -- Executing [...@pstn:1] DeadAGI(IAX2/#000105-4121, agi://localhost/ManageCalls.agi?when=after) in new stack -- AGI Script agi://localhost/ManageCalls.agi?when=after completed, returning 0 -- Hungup 'IAX2/#000105-4121' Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: HANGUP Timestamp: 01096ms SCall: 04121 DCall: 02203 [192.168.1.5:3048] CAUSE CODE : 16 Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 01096ms SCall: 02203 DCall: 04121 [192.168.1.5:3048] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 00452 DCall: 0 [192.168.1.35:4569] USERNAME: 400 REFRESH : 60
Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
On Tue, 15 Jun 2010, Faisal Hanif wrote: Till now I am not able to find any difference between both machines. Can you please tell me how I can try to resolve it on OS level using some utility like dig? I'm not a DNS guru, but this is what I would do... ) Start with whois to determine the names of the authoritative name servers for the domain: whois example.com This shows that A.IANA-SERVERS.NET and B.IANA-SERVERS.NET are the authoritative name servers. ) Query each of the name servers to see who they think are the name servers: dig @A.IANA-SERVERS.NET example.com ns dig @B.IANA-SERVERS.NET example.com ns This shows that a.iana-servers.net and b.iana-servers.net. The difference in case is not significant. ) Query each name server to see if they can resolve the host name: dig @a.iana-servers.net whsvoip.example.com dig @b.iana-servers.net whsvoip.example.com ) If you are trying to use SRV records, use these queries: dig @a.iana-servers.net srv _sip._udp.example.com dig @b.iana-servers.net srv _sip._udp.example.com ) And just for fun, repeat the whole exercise on the other host. You're looking for failures or inconsistencies that may explain why you getting different behaviors. -- 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 -- 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 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
Both have CentOS 5.2. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: Tuesday, June 15, 2010 11:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' What OS are you running on the two systems? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 15, 2010, at 10:40 AM, Faisal Hanif wrote: Till now I am not able to find any difference between both machines. Can you please tell me how I can try to resolve it on OS level using some utility like dig? Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller Sent: Tuesday, June 15, 2010 10:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' On Tue, Jun 15, 2010 at 09:35:37PM +0500, Faisal Hanif wrote: Hi, I am also wonder that same SRV record is working fine on one machine but not on 2nd while both have same asterisk version. It may be some missing OS utilities which asterisk using to resolve SRV? Could be. To test, does replacing whsvoip.globalipcom.com with, say, whs-proxy00.de.whsvoip.globalipcom.com (or whatever is reachable) make it work? What is different about the two machines you've tried? -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
One thing I would do is something like On system one: rpm -qa | sort sys1 On system two: rpm -qa | sort sys2 Then on either system do a diff of these two files. If you only use yum or rpm to install and update software you can tell what is different between the two systems. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 15, 2010, at 12:23 PM, Faisal Hanif wrote: Both have CentOS 5.2. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: Tuesday, June 15, 2010 11:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' What OS are you running on the two systems? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 15, 2010, at 10:40 AM, Faisal Hanif wrote: Till now I am not able to find any difference between both machines. Can you please tell me how I can try to resolve it on OS level using some utility like dig? Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller Sent: Tuesday, June 15, 2010 10:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' On Tue, Jun 15, 2010 at 09:35:37PM +0500, Faisal Hanif wrote: Hi, I am also wonder that same SRV record is working fine on one machine but not on 2nd while both have same asterisk version. It may be some missing OS utilities which asterisk using to resolve SRV? Could be. To test, does replacing whsvoip.globalipcom.com with, say, whs-proxy00.de.whsvoip.globalipcom.com (or whatever is reachable) make it work? What is different about the two machines you've tried? -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Cutting the CallerID(RDNIS)
What that catches though is the actual phone number not the RDNIS. Is
there any way that we can use something similar for sorting of the RDNIS?
On 6/15/2010 9:19 AM, Zeeshan Zakaria wrote:
Hi,
I do something similar but for outbound calls. But same should work
for inbound calls.
First catch the calls with a leading 1, then strip 1 and send them to
your desired context or extension. I use AEL, but in regular config it
should be something like:
[incoming-calls]
; catch numbers with leading 1 here
exten = _1X.,1,Goto(${EXTEN:1},1)
; numbers with no leading 1 go here
exten = _X.,1,Noop( - - - - - Incoming call - - - - -)
Zeeshan A Zakaria
--
www.ilovetovoip.com http://www.ilovetovoip.com
On 2010-06-15 9:27 AM, Chris Ramirez crami...@tele-onecom.com
mailto:crami...@tele-onecom.com wrote:
We are having an issue with the RDNIS coming through with a leading 1
on some calls. I have been trying to find a way to remove the leading
number only if it starts with a 1 and have yet to find a solid
solution. If anyone else has any idea as how to do this I would
greatly appreciate it! Thanks.
--
*Chris Ramirez*
TELE-ONE COMMUNICATIONS, INC.
crami...@tele-onecom.com mailto:crami...@tele-onecom.com
903-531-0777
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crami...@tele-onecom.com
903-531-0777
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Re: [asterisk-users] Unable to pickup an extension, tryi
Jonas,
Did you really read the whole wiki page ? I've not used PICKUPMARK
before myself, but if you want to pay someone consultant rates to do the
work for you, I'm sure there's plenty willing to take your money off
you. In the meantime, why dont you help yourself and really read the
whole page, including the many examples.
From your 1 line snippet of your dialplan (dont know how anyone is
supposed to be able to give you much help with such little info anyway),
I can only guess that since you are using the 's' extension, you are in
a macro ? If so, try scrolling down the wiki page to the example using
'[macro-inbound]'.
Rob
Jonas Kellens wrote:
Philipp,
I have read this wiki, but I still don't know how to implement it in
my example...
The wiki says :
exten = 1234,1,Set(__PICKUPMARK=1234)
So how do I do this with my dialplan :
exten = s,n,Dial(${SIPaccounts},${timeout})
or translated :
exten = s,n,Dial(SIP/testcorp1SIP/testcorp2)
Can you give me the exact syntax?? Because I really can not filter
this from the wiki.
Looking forward to your answer, thx !
Jonas.
On 06/15/2010 06:09 PM, Philipp von Klitzing wrote:
Hi!
How to do this ??
To proceed with your answer on PICKUPMARK, where do I put this ???
Look at the example for Asterisk 1.4 on this page:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
Philipp
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Re: [asterisk-users] Cutting the CallerID(RDNIS)
Can you give an example of how it looks like? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-15 4:18 PM, Chris Ramirez crami...@tele-onecom.com wrote: What that catches though is the actual phone number not the RDNIS. Is there any way that we can use something similar for sorting of the RDNIS? On 6/15/2010 9:19 AM, Zeeshan Zakaria wrote: Hi, I do something similar but for outbound c... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail vm-intro played even when temp greeting is setup
Hi there,
I am configuring a small voicemail server and I am facing the following
problem.
Executing this command: exten = 1234,1,VoiceMail(${numb...@test)
When a user does not have a customized temporary greeting vm-intro message
is played asking for the message to the user but when the user has already a
temporary greeting both the temporary greeting and vm-intro are played.
Basically what I would like to do is to avoid this second scenario so when a
user has a customized temporary greeting just that is played and not
vm-intro is played.
I have seen that to avoid the reproduction of vm-intro I can use the s flag,
doing something like this:
exten = 1234,1,VoiceMail(s${numb...@test)
But the problem is that if I do that nothing is played for the users that
don't have personalized greeting.
Any help would be appreciated.
Thanks in advance,
Jonathan
--
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Re: [asterisk-users] Voicemail vm-intro played even when temp greetingis setup
This should do the trick might have to change greet.WAV to some other
value
exten = 930,1,Answer
exten = 930,n,System(/bin/ls
/var/spool/asterisk/voicemail/default/${NUMBER}/greet.WAV)
exten = 930,n,verbose(returned ${SYSTEMSTATUS}
exten = 930,n,Gotoif($[${SYSTEMSTATUS} = SUCCESS]?play1)
exten = 930,n,Voicemail(s${numb...@test)
exten = 930,n,hangup
exten = 930(play1),n,Voicemail(${numb...@test)
exten = 930,n,hangup
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan
González
Sent: Tuesday, June 15, 2010 3:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail vm-intro played even when temp
greetingis setup
Hi there,
I am configuring a small voicemail server and I am facing the following
problem.
Executing this command: exten = 1234,1,VoiceMail(${numb...@test)
When a user does not have a customized temporary greeting vm-intro message
is played asking for the message to the user but when the user has already a
temporary greeting both the temporary greeting and vm-intro are played.
Basically what I would like to do is to avoid this second scenario so when a
user has a customized temporary greeting just that is played and not
vm-intro is played.
I have seen that to avoid the reproduction of vm-intro I can use the s flag,
doing something like this:
exten = 1234,1,VoiceMail(s${numb...@test)
But the problem is that if I do that nothing is played for the users that
don't have personalized greeting.
Any help would be appreciated.
Thanks in advance,
Jonathan
--
Personal webpage - www.jonbaraq.eu
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Re: [asterisk-users] Unable to pickup an extension, tryi
Rob, it's not a macro but a sub. In my previous post I posted more info, I am not going to post the whole output every time. I read on the wiki that you set the PICKUPMARK equal to the extension for that channel, but in my case I'm not using extensions but multiple SIPaccounts in my dial statement. I really see no example on the wiki on how to deal with multiple SIPaccounts/extensions in one dial()-statement... I ask for a clean example to show me how I need to implement it, not to do the whole writing of my dialplan. My dialplan consists of already +-2500 rules, no need for a consultant, I wrote it myself. Do you have another wiki ? Because even with the search-option I can not find the word inbound, as you refer to [macro-inbound]. I'm refering to http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup Jonas. On 06/15/2010 10:26 PM, Rob Coward wrote: Jonas, Did you really read the whole wiki page ? I've not used PICKUPMARK before myself, but if you want to pay someone consultant rates to do the work for you, I'm sure there's plenty willing to take your money off you. In the meantime, why dont you help yourself and really read the whole page, including the many examples. From your 1 line snippet of your dialplan (dont know how anyone is supposed to be able to give you much help with such little info anyway), I can only guess that since you are using the 's' extension, you are in a macro ? If so, try scrolling down the wiki page to the example using '[macro-inbound]'. Rob -- _ -- 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] Cutting the CallerID(RDNIS)
Right now we are attempting this... When a call comes in it has all of the CALLERID() information. What we are wanting is that when the variable CALLERID(RDNIS) comes through as 1800555 we can have it be forwarded through the system as 800555 rather than with the 1 preceding it.We are setting the RDNIS as the CDR(userfield) to pass it through. Is that what you were wanting? On 6/15/2010 3:23 PM, Zeeshan Zakaria wrote: Can you give an example of how it looks like? Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-06-15 4:18 PM, Chris Ramirez crami...@tele-onecom.com mailto:crami...@tele-onecom.com wrote: What that catches though is the actual phone number not the RDNIS. Is there any way that we can use something similar for sorting of the RDNIS? On 6/15/2010 9:19 AM, Zeeshan Zakaria wrote: Hi, I do something similar but for outbound c... -- _ -- 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 -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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] Extract user part from SIP URI
Hi there,
I am using the user part of a SIP URI to know the mailbox number that I have
to use on my dialplan. This SIP URI is stored in the History Info of the SIP
Header so I am extracting that user part right now using the CUT function
against that part of the header.
I would like to know if there's any simple method to get that part of the
SIP URI using regular expression or something easy and consolidated.
Right now what I am doing is (NUMBER contains the history info):
exten = 1234,1,Set(NUMBER=${CUT(NUMBER, \:, 2)})
exten = 1234,2,Set(NUMBER=${CUT(NUMBER, @, 1)})
I am just interested in the first occurrence of the SIP URI.
Thanks in advance,
Jonathan
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Personal webpage - www.jonbaraq.eu
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Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
On Tue, Jun 15, 2010 at 10:40:06PM +0500, Faisal Hanif wrote: Till now I am not able to find any difference between both machines. Can you please tell me how I can try to resolve it on OS level using some utility like dig? On Tue, Jun 15, 2010 at 09:35:37PM +0500, Faisal Hanif wrote: Hi, I am also wonder that same SRV record is working fine on one machine but not on 2nd while both have same asterisk version. It may be some missing OS utilities which asterisk using to resolve SRV? Could be. To test, does replacing whsvoip.globalipcom.com with, say, whs-proxy00.de.whsvoip.globalipcom.com (or whatever is reachable) make it work? What is different about the two machines you've tried? Hmm. I see something odd. My initial dig came back with a length of 515 bytes. Then: ; DiG 9.3.4 @auth210.ns.uu.net _sip._udp.whsvoip.globalipcom.com SRV +ignore [lines deleted] ;; SERVER: 195.129.12.74#53(195.129.12.74) ;; WHEN: Tue Jun 15 16:39:09 2010 ;; MSG SIZE rcvd: 457 Then: ; DiG 9.3.4 @auth210.ns.uu.net _sip._udp.whsvoip.globalipcom.com SRV [lines deleted] ;; SERVER: 195.129.12.74#53(195.129.12.74) ;; WHEN: Tue Jun 15 16:39:59 2010 ;; MSG SIZE rcvd: 515 The difference is that sometimes one more server is returned, pushing the response over 512 bytes. Now it's back to 457. I suspect that they know there's a problem, and are trying to figure out what to do about it. (maybe shorten their server names by one or two characters each?) So depending on when each of your boxes happens to issue the SRV query, they could very well be identical but get different results. I doubt it will help, but you can try quoting RFC 2782, Currently there's a practical limit of 512 bytes for DNS replies. Until all resolvers can handle larger responses, domain administrators are strongly advised to keep their SRV replies below 512 bytes. -- Barry -- _ -- 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] numbers
If i will use asterisk like a phone switch with voicemail What i need? Can i use a sip or voip account Like cellip.com Sorry my english -- _ -- 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] numbers
On Wed, 16 Jun 2010, mattias jonsson wrote: If i will use asterisk like a phone switch with voicemail What i need? Can i use a sip or voip account Like cellip.com A very simple question you could get the answer to immediately using a tool like Google. Try searching for something like asterisk first installation. After reading a couple of articles, if you still need help, please post your questions with more meaningful subjects -- better bait yields better fish. -- 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 -- 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] a2billing for residential voip usage
It was already done. My problem now is that I cant' place any calls through a2billing. --- On Tue, 6/15/10, Faisal Hanif fai...@vopium.com wrote: From: Faisal Hanif fai...@vopium.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:26 PM You need to copy or soft link a2billing.conf to /etc/ folder as by default latest version search for it in /etc/ Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Tuesday, June 15, 2010 9:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] a2billing for residential voip usage I copied the config to the a2billing.conf in /etc/asterisk folder. I'm still not able to place any calls yet. Looks like I have to read more on how to configure trunks and providers whick got me confused. I'll learn though. --- On Tue, 6/15/10, Vardan Harutyunyan hvarda...@gmail.com wrote: From: Vardan Harutyunyan hvarda...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 8:03 AM look manual, but in any case the a2billing.conf is in /etc/asterisk/ on can say, where you have place your asterisk configuration files -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Jimmy Godbout wrote: Hi, Maybe you can just use a reporting tool that will look at the CDR and tell you who's using the phone the most. Some of them will use a DB to store the CDR. If you want, you can even use Excel to look at the csv file created by default and make your own report. http://www.voip-info.org/wiki/view/Asterisk+billing http://www.voip-info.org/wiki/view/Asterisk+GUI (in Billing Call Detail Reporting) http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Jimmy -Original Message- From: landysacco...@yahoo.com Sent: Tue, 15 Jun 2010 00:11:51 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] a2billing for residential voip usage Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ramtalk2...@gmail.com wrote: From: ramtalk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landylandysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- 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 -Inline Attachment Follows- -- _ -- 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 Share photos screenshots in seconds... TRY FREE IM TOOLPACK at http://www.imtoolpack.com/default.aspx?rc=if1 Works
Re: [asterisk-users] Cutting the CallerID(RDNIS)
I have worked on somewhat similar situations but with ANI and DNIS, not with
RDNIS, but it seems similar to me. If it enters a context, the code I sent
above should work just fine. If not, you can either replace ${EXTEN:1} with
${CALLERID(RDNIS):1}, or within the same context where you know
CALLERID(RDNIS) exists assign it to a new variable foo like:
exten = s,1,GotoIf($[${CALLERID(RDNIS):0:1}=1]?a:b)
exten = s,n(a),Set(foo=${CALLERID(RDNIS):1})
exten = s,n(b),Noop( - - - - - RDNIS fix end - - - - -)
And from here you can use the foo variable. I don't know if it'll work or
not, but in the above code you can try to do:
Set(CALLERID(RDNIS)=${CALLERID(RDNIS):1})
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-15 5:13 PM, Chris Ramirez crami...@tele-onecom.com wrote:
Right now we are attempting this...
When a call comes in it has all of the CALLERID() information. What we are
wanting is that when the variable CALLERID(RDNIS) comes through as
1800555 we can have it be forwarded through the system as 800555
rather than with the 1 preceding it.We are setting the RDNIS as the
CDR(userfield) to pass it through. Is that what you were wanting?
On 6/15/2010 3:23 PM, Zeeshan Zakaria wrote:
Can you give an example of how it looks like?
...
--
Chris Ramirez
TELE-ONE COMMUNICATIONS, INC.
crami...@tele-onecom.com
903-531-0777
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[asterisk-users] Fwd: [INSTALL #RKZ-745226]: Digium Support Survey, Partial Faxes
The last time Digium gets a dime from me. -- Forwarded message -- From: Steve Totaro stot...@totarotechnologies.com Date: Fri, Jun 11, 2010 at 3:44 PM Subject: Re: [INSTALL #RKZ-745226]: Digium Support Survey, Partial Faxes To: supp...@digium.com Lets start with the stats first, this should be an easy one. The numbers below are from just today. How does it make sense? How can I create a report to give to the CTO without making up numbers? Connected to Asterisk 1.6.2.6 currently running on voipgw01 (pid = 2735) Verbosity is at least 3 voipgw01*CLI fax show stats voipgw01*CLI FAX Statistics: --- Current Sessions : 1 Transmit Attempts: 0 Receive Attempts : 25 Completed FAXes : 24 Failed FAXes : 5 voipgw01*CLI Digium G.711 Licensed Channels: 4 Max Concurrent : 0 Success : 0 Switched to T.38 : 0 Canceled : 0 No FAX : 0 Partial : 0 Negotiation Failed : 0 Train Failure: 0 Protocol Error : 0 IO Partial : 0 IO Fail : 0 Digium T.38 Licensed Channels: 4 Max Concurrent : 3 Success : 16 Canceled : 0 No FAX : 0 Partial : 2 Negotiation Failed : 0 Train Failure: 2 Protocol Error : 4 IO Partial : 0 IO Fail : 0 Thanks, Steve T On Fri, Jun 11, 2010 at 2:54 PM, Digium Install Support supp...@digium.comwrote: Hello, I would be glad to continue troubleshooting this issue with you. Could you please provide the fax debug output for a failed, and a successful fax? Regards, Derek Peloquin Digium, Inc. | Support Technician 2 dCAP – Digium Certified Asterisk Professional 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA support: +1 256-428-6161 local: +1 256-428-6000 toll free: +1 877-DIGIUM1 (344-4861) fax: +1 256-864-0464 Check us out at: www.digium.com www.asterisk.org Ticket Details === Ticket ID: RKZ-745226 -- _ -- 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] Small PC to build and run Asterisk
On Tue, 15 Jun 2010 07:58:34 -0400, SIP wrote: Danny Nicholas wrote: Also cheaper to replace flash card than hard drive. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Monday, June 14, 2010 4:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Small PC to build and run Asterisk Why no flash? * Small pre-built PC (not buying board, case, all parts separately) * Low power consumption * No fan or very small fan * Hard drive (not flash memory) An ssd uses less power, so generates less warmth, hence less need for fan in the drive area. Also less noise.. I like this one, or its smaller brother: http://www.fit-pc.com/web/fit-pc2/fit-pc2i-specifications/ But a flash card needs replacing more often than a hard drive. It's just not designed for the same sort of lifecycle of writes that a hard drive is. Sure, the number is always increasing as they increase the capacity, but it WILL NOT LAST. Dependent on the type of filesystem access you need, SSD could be a great choice. But if you're heavy on logging and writing small data bits here and there (which isn't always something you can control if you don't write all the software), then a hard drive is just going to be the better choice to hold up for a long period of time. This need not be the case. It depends upon what Asterisk distro you're using. I ran Astlinux from a vintage 256 MB CF card for several years without a problem. If you simply build up a server and use flash media in place of a disk then you will likely kill the media in a short period. The behaviour of the system needs to be tailored to running from Flash. Some distro's, like Askozia and Astlinux, have been specifically engineered around running from flash media. This basic form of operation has been well proven in projects like monowall and pfsense. For very large installations with a lot of I/O intensive extra activities running on the server running from flash may never be appropriate. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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
