[asterisk-users] OT: Cisco ATA 186
Hi all, do you know any firmware release which fixes that issue for cisco ATA186? ATA 186 3.x.x Cisco ATA 186 v3.* CANCEL requests can be sent with a completely bogus URI, making it impossible to cancel a call. bug no workarounds This is from: http://www.communigate.com/SIP/SIPProblems.html. In other words, you make a call, other side doesn't pickup, then you hang up, but destination phone still rings. Best regards, Sebastian -- _ -- 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] OT: NAT in SPA922
Ok.. here is how I solved. PC+IPPhone--Cisco2950Router. Each PC in one private subnet NATed on the router. All phones in same network (different from PCs). Sebastian On Fri, May 7, 2010 at 9:08 AM, James Lamanna jlama...@gmail.com wrote: On May 7, 2010, at 8:03, James Lamanna jlama...@gmail.com wrote: On Thu, May 6, 2010 at 8:14 PM, Vineet Bhojnagarwala vbho...@gmail.com wrote: Alternatively, if using normal vlans, this can also be achieved by enabling access list on the switch and restrict traffic flows. Generally this is done on a layer 3 switch, don't think it will support on your switch model. That is correct. In order to do this on a 2950, you will need a router behind this to be the gateway for each vlan. (On Cisco equipment you'd need to create a subinterface for each vlan (i.e. FastEthernet 0.xxx) where xxx is your vlan number. Then you can set each port up to be a trunk port on the 2950, but specify the native vlan on the port as the PC vlan # and allow the Vlan # for the phone vlan. So something like: switchport mode trunk switchport trunk native vlan [pc vlan #] switchport trunk allowed vlan [pc vlan #],[phone vlan #] Then you will have to create access-lists on the router to block intra-VLAN traffic. This can also be all done on a Layer 3 switch (like the Cisco 3550), by defining each VLAN as an interface: interface VLAN 100 description Phone VLAN ip address 192.168.100.1 255.255.255.0 ! interface VLAN 101 description Customer 1 VLAN ip address 192.168.101.1 255.255.255.0 ! etc.. then your ports will look like: interface FastEthernet 0/2 description customer 1 port switchport mode trunk switchport trunk encapsulation dot1q switchport trunk native vlan 101 switchport trunk allowed vlan 100,101 ! Then you'll need access lists to prevent the intra-vlan traffic.. I lied. You don't need access-lists in this case with the allowed vlan statement. -- James Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 7, 2010, at 8:39 AM, Vineet Bhojnagarwala vbho...@gmail.com wrote: I think this is a motel kind of situation and a PVLAN serves the situation right. Put all the ipphones in the voice vlan as suggested, make a seperate isolated vlan for the PCs, this will restrict traffic between the clients. Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 6, 2010, at 11:30 PM, David White david.wh...@watchguard.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Noah Miller Sent: Thu 5/6/2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: NAT in SPA922 It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. Take j's suggestion to use VLANs. This is not a good situation for NAT. Cisco 2950's can do VLANs. to be clear, the only way this will work with the PCs is if each PC vlan is *also* a unique ip subnet (else how do all the vlans access a common default gw?) place the phones in a voice vlan, and the phone problem is solved. as for the PC isolation, you might get better feedback on a cisco or other networking forum. -david -- _ -- 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
Re: [asterisk-users] OT: NAT in SPA922
Ok..So what ip phone model do NAT? Sebastian On Wed, May 5, 2010 at 12:26 PM, Luki lugos...@gmail.com wrote: However, when I connect a PC to that port, SPA922 works as bridge. Exactly. The SPA9x2 has a 2-port switch; no NAT, no routing (unlike the SPA2102, etc). I think the 5.1 series is the latest firmware for the 922; the the 942, there is 6.1.5a. Luki -- _ -- 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] OT: NAT in SPA922
It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Sebastian On Thu, May 6, 2010 at 12:36 PM, Noah Miller noahisaacmil...@gmail.comwrote: Ok..So what ip phone model do NAT? I think you'd struggle to find one. If it's a requirement you're probably doing something wrong... Definitely get a router. Plug the IP phone into the router, and then you can plug the computer into the phone or the router. - Noah -- _ -- 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] OT: NAT in SPA922
I see the following in SPA922 System tab (new firmware) VLAN Settings Enable VLAN:yesnoEnable CDP:yesno VLAN ID:PC Port VLAN Highest Priority:01234567No Limit Enable PC Port VLAN Tagging:yesnoPC Port VLAN ID: VLAN ID:1 for all Phones, and VLAN 2, 3, 4, 5..,24 for each PC. This should work, right? Sebastian On Thu, May 6, 2010 at 2:25 PM, Jeff LaCoursiere j...@jeff.net wrote: On Thu, 6 May 2010, Sebastian Milioto wrote: It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Sebastian Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. j -- _ -- 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] OT: NAT in SPA922
Hi all, I've just bought some SPA922. First time with this hardware for me. I see no LAN tab in its web GUI where I can setup NAT for PC conected to its LAN ethernet port. However, when I connect a PC to that port, SPA922 works as bridge. Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist such LAN tab for setting up parameters as port forwarding? (by the way, version is 5.1.15(a). I'll appreciate links for downloading new firmware) Thanks in advance, Sebastian -- _ -- 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] SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)
Thanks! On Thu, Mar 18, 2010 at 5:04 PM, Joseph syscon...@gmail.com wrote: On 03/18/10 16:22, Sebastian Milioto wrote: Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian You will find it here: http://prov.802.cz/fw/ Ever since the Linksys took over from Sipura and now by Cisco, thoese devices are of very poor quality. Two of SPA3102 died on me within two years, in addition lots of echo impossible to eliminate. I've switched/replaced my Linksys/Sipura units with Audiocodes MP-114 but they are not perfect either. Though, I can say they don't have/generate any echo problems and fixes go through without any problem (which I can not say the same about Linksys/Sipura units.) -- Joseph -- _ -- 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] SPA3102 + asterisk drop call and loop (was SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?) )
Ok, I downgraded spa3102 to 3.3.6. Now when I make a call from pstn and call is established asterisk seems to drop the call. However I still hearing ringback on pstn side, call is established again, and asterisk drops the call again, like a loop. -- Executing [preat_ad...@nodo:1] Playback(SIP/PSTN-08214948, horario-atencion/our-business-hours-are) in new stack -- SIP/PSTN-08214948 Playing 'horario-atencion/our-business-hours-are' (language 'es') == Spawn extension (nodo, preat_admin, 1) exited non-zero on 'SIP/PSTN-08214948' -- Executing [preat_ad...@nodo:1] Playback(SIP/PSTN-08214948, horario-atencion/our-business-hours-are) in new stack -- SIP/PSTN-08214948 Playing 'horario-atencion/our-business-hours-are' (language 'es') == Spawn extension (nodo, preat_admin, 1) exited non-zero on 'SIP/PSTN-08214948' -- Executing [preat_ad...@nodo:1] Playback(SIP/PSTN-08214948, horario-atencion/our-business-hours-are) in new stack -- SIP/PSTN-08214948 Playing 'horario-atencion/our-business-hours-are' (language 'es') == Spawn extension (nodo, preat_admin, 1) exited non-zero on 'SIP/PSTN-08214948' I've read this had happen to other people, however I can't find how they solved it. It seems to be a codec problem.. however I've already tried configuring g729a,g711u, and g711a in spa3102 with no success.. Can anybody help me with that, please? Sebastian On Fri, Mar 19, 2010 at 10:16 AM, Sebastian Milioto smili...@gmail.comwrote: Thanks! On Thu, Mar 18, 2010 at 5:04 PM, Joseph syscon...@gmail.com wrote: On 03/18/10 16:22, Sebastian Milioto wrote: Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian You will find it here: http://prov.802.cz/fw/ Ever since the Linksys took over from Sipura and now by Cisco, thoese devices are of very poor quality. Two of SPA3102 died on me within two years, in addition lots of echo impossible to eliminate. I've switched/replaced my Linksys/Sipura units with Audiocodes MP-114 but they are not perfect either. Though, I can say they don't have/generate any echo problems and fixes go through without any problem (which I can not say the same about Linksys/Sipura units.) -- Joseph -- _ -- 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] SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)
Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian -- _ -- 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] IP Phone recommendation
SPA504G seems to be a good choice since I don't want to have any issues and this still have the same provisioning scheme I'm working right now. Also I like the HD feature.. what is the bandwith consumption for that feature? It will work only between 2 spa504 right? or any G722 device can interoperate with it? Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP Phone recommendation
Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do you suggest according I said above? Is there another device which deserves attention? Thanks very much in advance, Sebastian Sebastian Milioto ITC Cid Campeadro 440 Rio Tercero, Cordoba, Argentina msn: sebamili...@hotmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone recommendation
I see... problem with spa941 is it dont have LAN port (I'm thinking NAT the customer PC) Sebastian On Wed, Feb 10, 2010 at 2:10 PM, Peter peterp...@aboutsupport.com wrote: On 10.2.2010 18:06, Steve Howes wrote: On 10 Feb 2010, at 15:50, Peder wrote: check out the Cisco SPA504G. They are the newer versions of the SPA922, support multiple lines and are fairly cheap too. I second that. They're rock solid, good audio quality and easy to provision. S SPA504G - 1 more vote for it. It is worth having 4 lines even if you need 1 initially. SPA504G supports G722 and sound is awesome even if you do not not use teh HD sound. If you do not care that mcuh about HD sound and do not need PoE SPA941 is a excellent choice - you get really a lot for the price Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alcatel OmniPCX Enterprise + Asterisk with E1
I see.. My hardware provider offers me TE122P with echo cancellation module VPMADT032. Would it be a good choice? Now, since my costumer already has its Alcatel PBX connected to E1 ILEC (with MFC-R2 signaling). Should I configure my Asterisk card to work with MFC-R2 for direct replacing?, or just configure the Alcatel PBX to work with default TEXXXP card configuration?. Is there any issues with MFC-R2 and Asterisk cards? Best Regards, Sebastian On Fri, Apr 17, 2009 at 4:48 PM, Jean-Denis Girard jd.gir...@sysnux.pfwrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Sebastian, Sebastian Milioto a écrit : | 1. Anybody has done this interconection? How does Asterisk and PBX | OmniPCX work together through an E1 interface? Any problems or bugs? I have two similar installations (OmniPCX-E1-Asterisk); they have been running, one of them since early 2005, passing more than 700 000 calls. | | 2. What E1 card should I buy for Asterisk? Is the physical interface | (conectors) E1 identical as T1? Digium TE110P or newer. | | 3. If cost wasn't a problem, do you suggest another interconection way | technically better? May be replacing Asterisk with another device with | an in-box E1? I now prefer to put asterisk between the telco and the PBX, using a dual E1 card. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkno3RoACgkQuu7Rv+oOo/iHZACfRSsheLqcnq5SDaiQkNs31Mfr cZYAoI5J6Bs8ld7nDG0IyFqrhFaubQ4w =51dT -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Alcatel OmniPCX Enterprise + Asterisk with E1
Hi all, I'm new in the forum, and although I have some experience in Asterisk, I've never work with Asterisk FXO, FXS, E1... cards. I have several costumers with ATAs working with my SER. However one of them bought an Alcatel PBX OmniPCX Enterprise and now he wants I give him a E1 interface for interconection with its new PBX. I understand I need a E1-IP gateway which could be Asterisk I think. So the network would look like this: Extensions-OmniPCX---E1---AsteriskIP--SER About that, I have a few question: 1. Anybody has done this interconection? How does Asterisk and PBX OmniPCX work together through an E1 interface? Any problems or bugs? 2. What E1 card should I buy for Asterisk? Is the physical interface (conectors) E1 identical as T1? 3. If cost wasn't a problem, do you suggest another interconection way technically better? May be replacing Asterisk with another device with an in-box E1? Thanks very much in advance, Sebastian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: SPA2100 transfer to ASTERISK CID
Are my e-mails arriving to the list? can somebody confirm? Sebastian -- Forwarded message -- Date: Fri, Nov 21, 2008 at 11:06 AM Subject: SPA2100 transfer to ASTERISK CID To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi all, I have around 100 SPA2100 registered in my provider openSER. I'd like to add an Asterisk registered into openSER, to the network, to deploy voicemail service for those SPAs. Due to administration access levels, I have no access to SER box, so I'm wondering if that possible: - Some foreign user (say A) calls one of my SPA (say B). - B don't answer. So.. B SPA is setted up to transfer in 20 seconds on no answer to the number in Asterisk. All ok so far, however, in Asterisk I receive de caller ID = A, but I need B CID. Having B caller ID I could let A leave the message into B mailbox. Can anybody helpme with that please? Thanks very much in advance Sebastian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA2100 transfer to ASTERISK CID
Hi all, I have around 100 SPA2100 registered in my provider openSER. I'd like to add an Asterisk registered into openSER, to the network, to deploy voicemail service for those SPAs. Due to administration access levels, I have no access to SER box, so I'm wondering if that possible: - Some foreign user (say A) calls one of my SPA (say B). - B don't answer. So.. B SPA is setted up to transfer in 20 seconds on no answer to the number in Asterisk. All ok so far, however, in Asterisk I receive de caller ID = A, but I need B CID. Having B caller ID I could let A leave the message into B mailbox. Can anybody helpme with that please? Thanks very much in advance Sebastian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ping
Ping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astbill DIALSTRING doesn't work
Hi all,
Im trying to setup Astbill in my Asterisk box, but I'm having some problems.
First I obtain the following when I want to make a call:
-- Executing AGI(SIP/71423-081b9010,
agiastar.agi|called_number) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agiastar.agi
-- SIP Seeding peer from astdb: '71423' at
[EMAIL PROTECTED]:5060 for 3600
-- AGI Script agiastar.agi completed, returning 0
-- Executing Dial(SIP/71423-081b9010, ) in new stack
Sep 4 15:23:25 WARNING[4224]: app_dial.c:781 dial_exec_full: Dial
requires an argument (technology/number)
== Spawn extension (mycontext, called_number, 3) exited non-zero on
'SIP/71423-081b9010'
-- Executing DeadAGI(SIP/71423-081b9010, agistardead.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agistardead.agi
-- AGI Script agistardead.agi completed, returning 0
-- SIP Seeding peer from astdb: '71423' at
[EMAIL PROTECTED]:5060 for 3600
In extensions.conf I have this:
exten = _XX,1,AGI(agiastar.agi,${EXTEN})
exten = _XX,2,Dial(${DIALSTRING})
I think I'm not getting nothing from DIALSTRING. How can I check that,
how can I resolve it?.
I tryed changing the second line bye for this:
exten = _XX,3,Dial(SIP/[EMAIL PROTECTED],90,Ttr)
and it starting work. At this point, I can make calls, but the billing
seems doesn't work. So, I think it is because I supressed the
DIALSTRING line.
Can anybody helpme with that? I cant find a manual or technical
information about astbill, except a few lines y Wiki. Is there
something like that?
Thanks very much in advance,
Sebastian
e-mail:[EMAIL PROTECTED]
msn:[EMAIL PROTECTED]
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[asterisk-users] astbill white screen!!
Hi all, I've installed asterisk and astbill according with all recommendation (mysql5, drupal included with astbill, php, apache2...). When I write http://server_adress/astbill, I get a white screen page. Browser doesn´t give me an error page, it just a white screen page. However asterisk doesnt have any problem, and works well with mysql. I also have installed Drupal 4.7.3 linked to other database with other user and password working well. And I have phpMyAdmin too. All working very good at the same server. I tried changing index.php to phpinfo.php in the same directory and it works well too. Can anybody help me with that please? Any suggestion will be very appreciated. Thanks, very much in advance Sebastian On 7/14/06, varun [EMAIL PROTECTED] wrote: Hello, Our asterisk server is on Centos 4.2 We want to use Astbill. Astbill requires Drupal and mysql 5. I could not find rpms mysql5 for centos. We are getting mysql extensions issues because of php-mysql. How do we solve this ? Any other billing software that similar to Astbill ? Thanks Varun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astbill white screen!!
Great help Time, thanks very much. That's the error I have Fatal error: SELECT command denied to user 'astbill_user'@'localhost' for table 'pbx_users' query: SELECT u.*, s.* FROM pbx_users u INNER JOIN pbx_sessions s ON u.uid = s.uid WHERE s.sid = '2e57e35eeb3e8464a4cc1bd10c1997a9' AND u.status 3 LIMIT 0, 1 in /home/astbill/wwwroot/includes/database.mysql.inc on line 66 I understand I have to GRANT all privileges to astbill_user over the asterisk database. Sebastian On 8/17/06, Time Bandit [EMAIL PROTECTED] wrote: I've installed asterisk and astbill according with all recommendation (mysql5, drupal included with astbill, php, apache2...). When I write http://server_adress/astbill, I get a white screen page. Browser doesn´t give me an error page, it just a white screen page. you have to enable it in php settings. Go in /etc/php.ini - change setting error_reporting to E_ALL - change setting display_errors to On - restart apache now, at least, it will tell you what goes wrong N.B.: display_errors should not be enabled on a production server hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom soundstation 501 crash
Hi all, I have a problem sending attached files with voicemail. I have postfix installed. When I write attach=no in voicmail.conf the notification is sent with no problem. But when I change to attach=yes, the notification never arrives. Could it be a postfix problem? Anybody could tell me how to configure it to permit attached files? I'm using mandriva 2006 Thanks very much in advance Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I can't call PSTN numbers
I run ngrep when I call an IP number and when I call a PSTN number, and the sequece is like that: For PSTN Numbers: Sipura --- Asterisk (Invite pstn number) Asterisk---Sipura(407 Proxy Auth. Required) Sipura --- Asterisk (Ack) Sipura --- Asterisk (Invite with Proxy Auth.) Asterisk---Sipura (100 Trying) Asterisk---SER (Invite pstn number) Asterisk---Sipura (180 Ringing) SER Asterisk (Trying) SER Asterisk (404 NOT FOUND) For IP Numbers: It is identical to the sequence above, but I get the following instead NOT FOUND: SERAsterisk (180 Ringing) SERAsterisk (200 OK) AsteriskSER (Ack) Asterisk---Sipura (OK) Sipura Asterisk (Ack) Then the call is established. So.. do you definitely think it is a SER configuration issue rather than Asterisk configuration issue? However, when I log in with a Sipura directly into SER, I can get access to all PSTN numbers. So why not with Asterisk?. I can't find anything different. Thanks again for your help Sebastian On 5/30/06, Woodoo People .pGa! [EMAIL PROTECTED] wrote: autocreatepeer=yes [ser_box1] type=peer username=ser_box1 insecure=yes canreinvite=no context=from-internal host=ip.address.of.box nat=yes disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm It doesn't work for me :-( How do you have the peer configuration in asterisk, to connect ot SER? exten = _4XX,1,Dial(SIP/[EMAIL PROTECTED]) it works to me (my provider sends me the last 3 digits) I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called [EMAIL PROTECTED] SIP/SER_ip_address-ec75 is circuit-busy Everyone is busy/congested at this time (1:0/1/0) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I can't call PSTN numbers
Hi all, I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called [EMAIL PROTECTED] SIP/SER_ip_address-ec75 is circuit-busy Everyone is busy/congested at this time (1:0/1/0) Then I hear busy tone. I tried thousands of parameters. (Ovbiously not the correct one :-s ) Any body could help me with this? Thanks very much in advance Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I can't call PSTN numbers
It doesn't work for me :-( How do you have the peer configuration in asterisk, to connect ot SER? Sebastian On 5/29/06, Woodoo People .pGa! [EMAIL PROTECTED] wrote: exten = _4XX,1,Dial(SIP/[EMAIL PROTECTED]) it works to me (my provider sends me the last 3 digits) I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called [EMAIL PROTECTED] SIP/SER_ip_address-ec75 is circuit-busy Everyone is busy/congested at this time (1:0/1/0) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_meetme.so
Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_meetme.so
Mmmm.. sorry... I'm rookie. Coudl you be a litlle more specific please? Where I have to uncomment ztdummy? On 4/13/06, Steve Totaro [EMAIL PROTECTED] wrote: Un-comment ztdummy and build re-zaptel and then re-build asterisk Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: Sebastian Milioto [mailto:[EMAIL PROTECTED] Sent: Thursday, April 13, 2006 10:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] app_meetme.so Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys pap2 behind Linksys RT31
Hi all, I have a public ip in Linksys RT31 (2 FXS port + 3 swtich port + 1 uplink port). I want to add behind it, a Linksys pap2 (uplink port + 2 FXS port) with private ip. I understand that I have to configure Port forwarding or port triggering (really I'm not sure which one). Is someone already configured this toplogy? Could you help me with that, please? Thanks very much in advance, Sebastian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with iptel.org
Hi all, I'm trying to connect my [EMAIL PROTECTED] to iptel.org, but the only I get is Allison telling me circuit busy now, please call again later or some thing similar. I'm trying make it by AMP and editing sip.conf and extension.conf, and I read all about it in voip-info.org. I will appreciate your help, Thanks in advance, Sebastian e-mail:[EMAIL PROTECTED] IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk with iptel.org
With sip show registry I can see I my asterisk is registered:
Host: iptel.org:5060
Username:84565616
Refresh: 145
State: Registered
The following is part of sip.conf:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=g729
allow=alaw
allow=ulaw
context = from-sip
callerid = Unknown
register = 84565616:[EMAIL PROTECTED]/200 ;iptel register
[200]
username=200
type=friend
secret=otro_password
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=Sebastian 200
[iptel]
type=friend
username=84565616
secret=password_iptel
fromdomain=iptel.org
host=iptel.org
And the following is part of extensions.conf
[outbound-allroutes]
include = outbound-allroutes-custom
include = outrt-001-llamadasSIP
[outrt-001-llamadasSIP]
include = outrt-001-llamadasSIP-custom
;exten = _9.,1,Macro(dialout-trunk,2,${EXTEN:1},00323) before,
this line was uncommented and the following line doesn't existed. But
it doesn't work, so I changed it by the following line
exten = _9.,1,Macro(dialout-SIPiptel,2,${EXTEN:1},00323)
exten = _9.,2,Macro(outisbusy) ; No available circuits
[macro-dialout-SIPiptel]
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) ;To take the
SIP trunk I dial 9
exten = 200,1,Aswer
exten = 200,3,Hangup
; dialout using a trunk, using pattern matching (don't strip any prefix)
; arg1 = trunk number, arg2 = number, arg3 = route password
[macro-dialout-trunk]
exten = s,1,GotoIf($[foo${ARG3} = foo]?3:2)) ; arg3 is pattern password
exten = s,2,Authenticate(${ARG3})
exten = s,3,Macro(record-enable,${CALLERIDNUM},OUT)
exten = s,4,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?7) ;check for
CID override for exten
exten = s,5,SetCallerID(${ECID${CALLERIDNUM}})
exten = s,6,Goto(9)
exten = s,7,GotoIf($[foo${OUTCID_${ARG1}} = foo]?9) ;check for CID
override for trunk
exten = s,8,SetCallerID(${OUTCID_${ARG1}})
exten = s,9,SetGroup(OUT_${ARG1})
exten = s,10,CheckGroup(${OUTMAXCHANS_${ARG1}})
; if we've used up the max channels, continue at 109 (n+101)
exten = s,11,SetVar(DIAL_NUMBER=${ARG2})
exten = s,12,SetVar(DIAL_TRUNK=${ARG1})
exten = s,13,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the
proper dial string for this trunk
exten = s,14,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ;
OUTNUM is the final dial number
exten = s,15,Cut(custom=OUT_${ARG1},:,1) ; Custom trunks are
prefixed with AMP:
exten = s,16,GotoIf($[${custom} = AMP]?19)
exten = s,17,Dial(${OUT_${ARG1}}/${OUTNUM}) ; Regular Trunk Dial
exten = s,18,Goto(s-${DIALSTATUS},1)
; This is a custom trunk. Substitute $OUTNUM$ with the actual number
and rebuild the dialstring
; example trunks: AMP:CAPI/:b$OUTNUM$,30,r,
AMP:OH323/[EMAIL PROTECTED]:
exten = s,19,Cut(pre_num=OUT_${ARG1},$,1)
exten = s,20,Cut(the_num=OUT_${ARG1},$,2) ; this is where we expect
to find string OUTNUM
exten = s,21,Cut(post_num=OUT_${ARG1},$,3)
exten = s,22,GotoIf($[${the_num} = OUTNUM]?23:24) ; if we didn't find
OUTNUM, then skip to Dial
exten = s,23,SetVar(the_num=${OUTNUM}) ; replace OUTNUM with the
actual number to dial
exten = s,24,Dial(${pre_num:4}${the_num}${post_num})
exten = s,25,Goto(s-${DIALSTATUS},1)
exten = s,111,Noop(max channels used up)
exten = s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten = s-BUSY,2,Busy()
exten = s-BUSY,3,Wait(60)
exten = s-BUSY,4,NoOp()
Hope you can help me. I'm afraid I make more trouble trying to fix it
Regards,
Sebastian
___
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Re: [Asterisk-Users] SIP audio port usage
But, if I have Xlite running on client PC and at the same time the user is doing FTP, both service has the same QoS treatment? Is there a way to differentiate these services besides the port? Sebastian On 9/20/05, Sherwood McGowan [EMAIL PROTECTED] wrote: Yes, because then the MACs specified would be getting the QoS, not just certain ports. This is how I set up my customers when they have QoS available. --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Adrien Laurent -Sent: Tuesday, September 20, 2005 8:53 AM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] SIP audio port usage - -So the more reliable way to do QoS is with MAC adress and not -on a port basis. -Am I right ? - -Thanks for your help, - -Adrien - -On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: - - I know that SIP is using port 5060 for session -initiation, but which - port does it use for audio ? is it dynamically assigned ? - - Its dynamically assigned on a per-call basis. - - Asterisk assigns the port based on contents of rtp.conf. - - Remote sip phones assign port numbers based on whatever the - manufacturer happened to choose (no industry standard). E.g., Cisco - uses 32,768 to something around 40,000, while xlite uses -something in the area of 8,000. - The various manufacturers are not consistent at all. - - - - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - - - --- -Adrien Laurent -[EMAIL PROTECTED] -www.modulis.ca -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk in Spanish
Hi all, I've been installing [EMAIL PROTECTED] and (of course) all the answering machine (I don't sure that's the right word in english, preatendedora in spanish) speech is in enlgish languaje. Is there anyway to download all those .gsm files speaked in spanish? Or may be another site which contain this kind of stuff (.wav, .gsm files for answering machines in spanish)? Thank you very much, Regards, Sebastian Milioto Telecommunications Engineer IM: [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] Mobile: 549 3571 543658 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
