RE: [Asterisk-Users] fedora core 3 kernel source - could someone throw the dog a bone!
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Harald Holzer Sent: Wednesday, August 24, 2005 3:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] fedora core 3 kernel source - could someone throw the dog a bone! I know this is a question with an obvious answer to some, but I am not one of them. Installed FC3, but this time I decide to update since my ISOs are a bit old, so typical yum update Downloaded the FC3 SRPM for my kernel 2.6.12... Installed the SRPM package Ran rpmbuild -bp -target=i686 kernel-2.6.spec why to recompile the kernel package ? Because I don't know any better? Do I even need the kernel source RPM or just kernel-smp-devel? When making zaptel; Make clean Make linux26 -- is this still required in current CVS head? Make install Tried to build zaptel - error; You do not appear to have the sources for the 2.6.12-1.1372_FC3smp kernel installed. yum install kernel-smp-devel ln -s /lib/modules/`uname -r`/build/ /usr/src/linux-2.6 should help ;-) Yes it did, and I thank you! Damon ___ --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
RE: [Asterisk-Users] Asterisk set-up for LCR
Why don't you post YOUR config files, then you might get some replies as to what is wrong. What you are trying to do can be done. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Huw Morgan Sent: Tuesday, August 23, 2005 8:33 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk set-up for LCR Hi, This is what I want to do: 1. Asterisk to answer calls via DID's, currently using SIPGATE 2. Provide a menu, and allow users to dial out. 3. According to the country and area they dial, the call should connect via one of up 4 carriers depending on cost. 4. If the carrier is busy it should go to the next one in line and so forth. I have tried to set this up, but it never answers the calls from Sipgate via SIP. Has anyone done anything like this before, and if so can they PLEASE give me some pointers or conf files so that I can see where I am going wrong on this!!! Thanks Huw. _ Be the first to hear what's new at MSN - sign up to our free newsletters! http://www.msn.co.uk/newsletters ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] any ISDN/PRI signaling experts out there?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul Belanger Sent: Friday, August 19, 2005 4:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] any ISDN/PRI signaling experts out there? See comments inline! Damon Estep wrote: I have officially engaged in a pissing contest with the local Telco over PRI calling name delivery. Welcome to my world, I deal with theses guys daily! Errgiant arn't they. We have a saying around work 'The telco is always wrong!'. The telco publishes their calling name delivery over PRI feature as being bellcore gr-1367-core compliant. The gr-1367-core spec states that the calling name is to be included as a facility IE in the setup message, or sent in a subsequent facility IE message with an indicator in the setup message that the CNAM will follow. Extensive testing and ISDN/PRI protocol analysis shows that the facility IE they are sending out with the CNAM in it comes only after we have sent back PROGRESS and ALERTING in response to the SETUP. If we block the PROGRESS and ALERTING and sit and WAIT for the FACILITY we never get it, the call will time out, so we know they are actually waiting for the call to progress before sending the facility IE CNAM. This sounds a little fishy, Orgination Number is usually transmitted in the SETUP message. Your are almost correct in your messaging: Network User(Switch) Setup CALL PROCEEDING ALERTING CALL CONNECT CALL CONNECT ACKNOWLEDGE There is about a 4sec timeout allow after SETUP is initially sent, if CALL PROCEEDING is not transmitted by that time, the Network side will terminiate the call. As far as I can tell the GR-1367-CORE spec does not define a maximum delay in sending the facility IE or whether it is acceptable to wait for PROGRESS and ALERT before sending it. The setup is; Telco PRI Lucent 5ESS Lucent MAX TNT Asterisk Here is an ISDN trace from a Dialogic board attached to a 5ESS switch with framing/coding ESF/B8ZS: SETUP(0x05) 1: BEARER CAPABILITY(0x04) 2: IE Length(0x03) 3: 1--- Extension Bit -00- Coding Standard ---0 Info. Transfer Cap. 4: 1--- Extension Bit -00- Transfer Mode ---1 Info. Transfer Rate 5: 1--- Extension Bit -01- Layer 1 Indent ---00010 User Info. Layer 1 1: CHANNEL ID(0x18) 2: IE Length(0x03) 3: 1--- Extension Bit -0-- Interface ID Present --1- Interface Type ---0 Spare 1--- Preferred/Exclusive -0-- D-Channel Indicator --01 Info. Channel Sel. 3.2: 1--- Extension Bit -00- Coding Standard ---0 Number Map 0011 Channel/Map Element 4: 1--- Extension Bit -001 Channel Number/Slot Map 1: CALLING PARTY NUM(0x6c) 2: IE Length(0x0b) 3: 1--- Extension Bit -010 Type Of Number 0001 Numbering Plan ID 949459 Number Digit(s)-- Here is the ANI 1: CALLED PARTY NUM(0x70) 2: IE Length(0x04) 3: 1--- Extension Bit -100 Type of Number 0001 Numbering plan ID 200 Number Digit(s) -- Here is the DNIS Notice my comments on where ANI and DNIS arrive in the SETUP message. The MAX TNT responds to the Facility IE with ISDN error 98, invalid message for call state. This is an actual CAUSE CODE from Q.931: Cause No. 98 - Message not compatible This cause indicates that the message received is not compatible with the call state or the message type is non-existent or not implemented. In short it is a protocol error. Check out http://www.telos-systems.com/?/techtalk/cause.htm for a complete lists of causes and there meaning. The SIP INVITE from the TNT to Asterisk contains no Caller Name information. It seems really odd to me that a Lucent TNT can not translate the caller ID Name info delivered by a Lucent 5ESS switch. On the same setup, if I connect another PRI device to it that emulates switch side signaling and includes the CNAM as a Display IE in the setup, the SIP invite is properly formatted and * receives the calling party name. Does anyone here have enough experience with ISDN PRI signaling to comment with some level of authority on this? Can you set a ISDN trace from your telco to your switch? I would be curious to see what it looks like. Again, it looks like your telco's problem. Your best to ask them to through a ThunderBird (T-Bird) on your circuit at your demarc and ask them if they see the CallerID, chances are they don't Damon Peter, Keep in mind it is CALLER ID NAME
[Asterisk-Users] any ISDN/PRI signaling experts out there?
I have officially engaged in a pissing contest with the local Telco over PRI calling name delivery. The telco publishes their calling name delivery over PRI feature as being bellcore gr-1367-core compliant. The gr-1367-core spec states that the calling name is to be included as a facility IE in the setup message, or sent in a subsequent facility IE message with an indicator in the setup message that the CNAM will follow. Extensive testing and ISDN/PRI protocol analysis shows that the facility IE they are sending out with the CNAM in it comes only after we have sent back PROGRESS and ALERTING in response to the SETUP. If we block the PROGRESS and ALERTING and sit and WAIT for the FACILITY we never get it, the call will time out, so we know they are actually waiting for the call to progress before sending the facility IE CNAM. As far as I can tell the GR-1367-CORE spec does not define a maximum delay in sending the facility IE or whether it is acceptable to wait for PROGRESS and ALERT before sending it. The setup is; Telco PRI Lucent 5ESS Lucent MAX TNT Asterisk The MAX TNT responds to the Facility IE with ISDN error 98, invalid message for call state. The SIP INVITE from the TNT to Asterisk contains no Caller Name information. It seems really odd to me that a Lucent TNT can not translate the caller ID Name info delivered by a Lucent 5ESS switch. On the same setup, if I connect another PRI device to it that emulates switch side signaling and includes the CNAM as a Display IE in the setup, the SIP invite is properly formatted and * receives the calling party name. Does anyone here have enough experience with ISDN PRI signaling to comment with some level of authority on this? Damon ___ 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] options for mysql query from dialplan
I am using realtime mysql for extensions, sip, and voicemail. Outbound call routing does not really perform well in realtime extensions due to the high number of rows in the database (300k), so I can not use it. It appears with my limited knowledge that the query method is not robust enough for large databases. Given the fact that I already have realtime and mysql configured, what are my options for running a mysql query from the dialplan to find the provider I want to use for outbound. I am not looking for a complete solution, just a hint on the best way to query my existing mysql database from the dialplan. I have looked at the MySQL command, and there are a lot of notes about connection closing and other scary stuff? Does it work? Are there other native options given the fact that realtime is configured and in use? The goal is to run a query against a database like this SELECT provideralias FROM ldproviders WHERE npa = (digits 2 thru 4 of dialed number) AND nxx = (digits 5 thru 7) Then take the provider alias returned and Dial(SIP/[EMAIL PROTECTED],60). Next step would be to add a loop for multiple providers, starting with the lowest cost. Any hints or comments from the pros? TIA Damon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] options for mysql query from dialplan
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roger Gulbranson Sent: Thursday, August 18, 2005 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Roger Gulbranson Subject: Re: [Asterisk-Users] options for mysql query from dialplan On Thu, 2005-08-18 at 09:41 -0600, Damon Estep wrote: I am using realtime mysql for extensions, sip, and voicemail. Outbound call routing does not really perform well in realtime extensions due to the high number of rows in the database (300k), so I can not use it. It appears with my limited knowledge that the query method is not robust enough for large databases. Given the fact that I already have realtime and mysql configured, what are my options for running a mysql query from the dialplan to find the provider I want to use for outbound. I am not looking for a complete solution, just a hint on the best way to query my existing mysql database from the dialplan. I have looked at the MySQL command, and there are a lot of notes about connection closing and other scary stuff? Does it work? Are there other native options given the fact that realtime is configured and in use? The goal is to run a query against a database like this SELECT provideralias FROM ldproviders WHERE npa = (digits 2 thru 4 of dialed number) AND nxx = (digits 5 thru 7) Then take the provider alias returned and Dial(SIP/[EMAIL PROTECTED],60). Next step would be to add a loop for multiple providers, starting with the lowest cost. Any hints or comments from the pros? Have you added appropriate indexes to your tables? ___ Yep, but all the indexing in the world is not going to change the fact that realtime extensions pulls ALL records in where the context matches, the priority is 1, and the extension starts with an underscore! We have over 100k extension in one table that start with an _ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] options for mysql query from dialplan
Could we not do away with PHP and AGI if realtime extensions had the ability to extend the pattern match query from _ to _ plus (n) number of dialed digits from the left? Damon -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Thursday, August 18, 2005 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] options for mysql query from dialplan Hi Damon, You are basically doing EXACTLY what we are doing right now; except we are doing more. We now have an AGI PHP script that does the following for every call: - Connect to MySQL over LAN - If the dialed number begins with 1, strip it. - SELECT State FROM lcr_lata WHERE NPA = $dial_npa AND NXX = $dial_nxx - Do some PHP logic to determine if Interstate vs Intrastate - SELECT rate, address, technology, prefixes FROM lcr_rates LEFT JOIN lcr_carriers USING(carrierid) WHERE NPA = $dial_npa AND NXX = $dial_nxx AND carrier_active = 1 ORDER BY rate ASC; - Loop thru results. lcr_rates has 329,530 rows. lcr_carriers has 8 rows. lcr_lata has over 150,000 rows. Everything preforms in real time. Here is a sample query of a call that just went thru: SELECT r.Interstate, rc.name, rc.technology, rc.address, rc.prefix FROM lcr_rates r LEFT JOIN lcr_carriers rc ON r.CarrierId = rc.id WHERE r.NPA = '254' AND r.NXX = '463' AND r.active = 1 ORDER BY r.Intrastate ASC, r.NPA DESC, r.NXX DESC Query took 0.0025 sec. I don't see how your table with 300K rows is preforming worse than ours. You got indexes? To make this even better, our MySQL server is a Quad P3 500 Mhz machine. Works great here. -Matthew Damon Estep wrote: I am using realtime mysql for extensions, sip, and voicemail. Outbound call routing does not really perform well in realtime extensions due to the high number of rows in the database (300k), so I can not use it. It appears with my limited knowledge that the query method is not robust enough for large databases. Given the fact that I already have realtime and mysql configured, what are my options for running a mysql query from the dialplan to find the provider I want to use for outbound. I am not looking for a complete solution, just a hint on the best way to query my existing mysql database from the dialplan. I have looked at the MySQL command, and there are a lot of notes about connection closing and other scary stuff? Does it work? Are there other native options given the fact that realtime is configured and in use? The goal is to run a query against a database like this SELECT provideralias FROM ldproviders WHERE npa = (digits 2 thru 4 of dialed number) AND nxx = (digits 5 thru 7) Then take the provider alias returned and Dial(SIP/[EMAIL PROTECTED],60). Next step would be to add a loop for multiple providers, starting with the lowest cost. Any hints or comments from the pros? TIA Damon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk command realtime
Anyone know if the application command Realtime() in asterisk can do more complex queries, like match the values in 2 columns? Show application realtime suggests it might be limited to one parameter queries. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail Retrival
There is a different approach to this;
Put a priority 'a' in the extension dialplan that goes to
Voicemmailmain(${EXTEN})
Users then dial there own extension from any location and press the *
key once voicemail picks up.
This method seems to emulate what most people are already used to.
If you have a voicemail button on the phone the other method works as
well, you can use both.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Derek Whitten
Sent: Wednesday, August 17, 2005 8:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail Retrival
you could declare the phone names as variables..
PHONE1=SIP/phone1
PHONE1VM=12345
On Wed, 2005-08-17 at 03:31, Rudolf Ladyzhenskii wrote:
Hi,
This procedure will work under one condition -- your user names are
same as your extension numbers. I have same problem. I was giving
phones alphanumeric user names, like phone1.
When VoicemailMain is called with ${CALLERIDNUM}, it is actually
called as VoiceMailMain(phone1). As a result, voice mail is asking
for a mailbox number which is same as your extension number. (BTW,
is
there a way to extract extension number rather than phone name?).
As I am experimenting with *, I will rename phones to match their
extensions.
Rudolf
- Original Message -
From: Sharadindu Mohanty
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, August 17, 2005 8:32 PM
Subject: Re: [Asterisk-Users] Voicemail Retrival
I did the same way but it is asking for some password and
mailbox. I think mail box is extension no but what abt
password?
Can i overide this procedure?
Thanks
Christoph Eicke [EMAIL PROTECTED] wrote:
On Wednesday 17 August 2005 10:29, Sharadindu
Mohanty
wrote:
Hi,
Hi!
Any ideas??
Yes, I do it in the following way. In extension.conf
add this line:
exten = ,1,VoiceMailMain(s${CALLERIDNUM})
exten = ,2,Hangup()
Here any extension can call and then
automatically gets directed to their
voicemail where they have some options.
I hope this helps,
Christoph
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RE: [Asterisk-Users] realtime caching
It seems that some options are not re-read when caching is on, for example, changing the caller ID value in the sip table has no effect until a reload (or expiration), so at least in some cases rtcahcefriends makes realtime notsorealtime. No. It is doing exactly what it says it will, cacheing. If you have rtcachefriends turned on, when a peer/user registers the info is pulled from DB and added to the internal (a la 'in memory') list that chan_sip maintains. If you change something in DB after this occurs then your changes won't take affect because chan_sip has no need to re-lookup your phones info since the info is already present in memory. What you can do is use sip prune realtime name to remove just the single peer/user from memory. And you can force a reload of that peer from realtime by using sip show peer name load. If you want pure realtime where chan_sip always pulls from db, then turn caching off. Keep in mind that turning caching off will remove MWI and NAT functionality. -Matthew What would it take (you, $) to add functionality that is a cross between caching and not, that is it caches with a flag in the extension, so if the flag is present realtime will be queried even though the extension is in cache when a new call comes IN TO that extension. Outgoing calls would not really need a re-query unless something about the provisioning of the phone changes, at which point it would re-register anyways, right? The goal is caching for MWI and NAT but realtime for calling, so the database is checked on every inbound call in case the dialplan changed, and the cache updated accordingly. Maybe a TTL flag, and when the TTL expires the cache entry stays, but is re-queried when a dialplan match is found. The admin could then tune the performance by setting different TTLs, maybe 15 minutes for lightly loaded systems, 4 hours for heavy loaded systems. Dynamic updates take place in whatever timeframe is specified on the TTL or less. Have I missed something, is this functionality already present? Damon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] realtime caching
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Wednesday, August 17, 2005 9:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] realtime caching It seems that some options are not re-read when caching is on, for example, changing the caller ID value in the sip table has no effect until a reload (or expiration), so at least in some cases rtcahcefriends makes realtime notsorealtime. No. It is doing exactly what it says it will, cacheing. If you have rtcachefriends turned on, when a peer/user registers the info is pulled from DB and added to the internal (a la 'in memory') list that chan_sip maintains. If you change something in DB after this occurs then your changes won't take affect because chan_sip has no need to re-lookup your phones info since the info is already present in memory. What you can do is use sip prune realtime name to remove just the single peer/user from memory. And you can force a reload of that peer from realtime by using sip show peer name load. If you want pure realtime where chan_sip always pulls from db, then turn caching off. Keep in mind that turning caching off will remove MWI and NAT functionality. -Matthew What would it take (you, $) to add functionality that is a cross between caching and not, that is it caches with a flag in the extension, so if the flag is present realtime will be queried even though the extension is in cache when a new call comes IN TO that extension. Outgoing calls would not really need a re-query unless something about the provisioning of the phone changes, at which point it would re-register anyways, right? The goal is caching for MWI and NAT but realtime for calling, so the database is checked on every inbound call in case the dialplan changed, and the cache updated accordingly. Maybe a TTL flag, and when the TTL expires the cache entry stays, but is re-queried when a dialplan match is found. The admin could then tune the performance by setting different TTLs, maybe 15 minutes for lightly loaded systems, 4 hours for heavy loaded systems. Dynamic updates take place in whatever timeframe is specified on the TTL or less. Have I missed something, is this functionality already present? Damon ___ I may have answered my own question, is it true that realtime extensions are still queried every call, and only chan_sip is effected by rtcachefriends? Damon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail crashes asterisk
It was fixed a while ago, download new code. There is a bug in the tracker on it. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Wednesday, August 17, 2005 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Voicemail crashes asterisk When a user dial voicemail and just hangs up or enters the wrong password 3 times asterisk will crash. We are using Cisco 7960G with SIP My asterisk is CVS-HEAD built on 2005-08-02 23:47:59 UTC Any help would be great!!! Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and LCR
Hello, How do you guys implement LCR in Asterisk? I have experimented with 2 ways, both seem to have issues and further testing is taking place now. Method1, use realtime for extensions and load your routing tables in an outbound context. Our requirements are LCR for the ~150,000 USA NPA-NXX combinations, everything outside of that goes through a single carrier so no routing needed. The performance stinks, takes too long to start the call. I need to do further testing and see if this is just a MySQL server (hardware) performance issue or a database structure issue (missing useful indexes). The second method is to use #include lcrtableflatfile.conf in extensions .conf and drop a single outbound context in that file with all of the routes. This method is far faster completing calls, but the asterisk reload command takes a long, long, long time (several minutes) to read the huge file. During the reload calls fail. Any input from others that have already done what I am doing would be helpful, what works best? Damon ___ 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] quad t1 / 1U rack server combos
It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. Cmon guys! Certify a few current model servers and be done with it. Without that information I must again ask the question; What 1u server combos work with the new quad pri cards UNDER LOAD (more than 75% channel use). Every user that buys a Digium PRI card should not have to play hit or miss with 2 or 3 servers that cost more than the card to get it to work. Please Please Please publish something useful to support the sale of PRI cards. Damon ___ 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] calling number type
Is there a method in SIP to set the CALLING number type to national and the calling number plan to isdn? I am dealing with an issue where a media gateway is not sending the correct values and would like to know if SIP has an equivalent parameter that can be set and mapped in the media gateway sip-isdn translations. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and LCR
How about this: 1. Put all the routes of all the providers in a MySQL table 2. Write a script with a 'clever' algorithm to find out cheapest route of each prefix. Are you saying realtime mysql is not clever? That is exactly what it is supposed to do. 3. Based on #2.. make a lcr_cheapest_route.conf 4. include lcr_cheapest_route.conf in extension.conf That is what we are doing, 3 minute reloads! But I don't know, how much resource asterisk will take after loading lcr_cheapest_route.conf Also, I don't have any idea about the performance would be. What do you think? Thanks -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 16 Aug 2005 12:57:14 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk and LCR On Tue, Aug 16, 2005 at 10:22:01AM -0600, Damon Estep wrote: Any input from others that have already done what I am doing would be helpful, what works best? For 100k routes+, you will have trouble holding them in a SQL database, particularly if your route selection query is complex. With a modern PC running PostgreSQL, you'll run into trouble at around 250k BHCA even with a much smaller number of routes. (This is quite apart from Asterisk itself, try writing a simple program that runs sample queries in a loop, perhaps with several threads. To a certain extent it depends on how you write the query and how judiciously you place indexes on the tables) When you want NPANXX granularity from several carriers (commonly 75-100k routes each) you'll get hit even worse. In my experience the safe limits of this approach are about a 2x DS3 worth of traffic with 10,000 routes in the table... After that you've got to pull everything into RAM and write a clever route selection algorithm... -w -- William Waites ww [EMAIL PROTECTED] magicphone.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quad t1 / 1U rack server combos
Damon Estep wrote: What 1u server combos work with the new quad pri cards UNDER LOAD (more than 75% channel use). Every user that buys a Digium PRI card should not have to play hit or miss with 2 or 3 servers that cost more than the card to get it to work. We use a Sangoma 4 port T1 card in our Dell Poweredge 1850 (1U) and it works like a champ. -Matthew One of the obvious disadvantages in using Sangoma cards would be Marksters interest is supporting them, using a TNT right now, and there are minor caller ID issues. The whole idea is to use a card offered by the company managing the project so interoperability is almost guaranteed. With that aside, what are the other pros/cons of the sagnoma cards? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and LCR
Are you saying realtime mysql is not clever? That is exactly what it is supposed to do. BTW, how do you integrate mysql with asterisk? any link, documention, tutorials would be greatly helpful. Search www.voip-info.org for asterisk realtime ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quad t1 / 1U rack server combos
What does the foneBRIDGE do that a Lucent TNT won't? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Tuesday, August 16, 2005 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] quad t1 / 1U rack server combos William - You should take a look at the foneBRIDGE, new product from redFONE. It has (4) PRI interfaces, and you run our to your primary and failover Asterisk servers via Ethernet. It does not do load balancing. but if you have a hardware failure in your primary Asterisk box, you can just fail right over to your secondary box. You don't need any PRI interface cards in your Asterisk host server at all. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory William Boehlke wrote: In our opinion, BAD idea to put four T1s on a single box, unless you have another box that also has 4 T1s. When, not if, the board fails, you have to take your box down to replace it. And as with anything having to do with computers you are guaranteed a failure at a peak time. Better to split the load between two boxes. William Boehlke Signate *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Chad Osmond *Sent:* Tuesday, August 16, 2005 12:15 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [Asterisk-Users] quad t1 / 1U rack server combos From what I understand (From Sangoma's tech support) and having a IBM x306 SCSI system with an A102u I believe that the system will scale up to 4xT1's easily. With a full T1 of traffic coming in and playing music on hold, the CPU was at 7% with no transcoding. Sangoma cards are supposed to place less draw on the interrupts and offer some new direct writing to DMA in their A104 cards. You may want to give them a call (Scott or Nenad are the two best people to speak with). From Sangoma README.asterisk: * Voice data is channelized and grouped into 8 byte chunks in HARDWARE. Each voice channel is then DMAed directly into the ZAPTEL buffers. Thus there is ZERO copy from HARDWARE to ZAPTEL, resulting in better performance and scalability.* It sounds to me like that would be once advantage over Digiums cards. They also have Hardware PRI functions that are passed directly to libpri. http://sangoma.com/linux/README.asterisk Hope that helps. Chad *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Damon Estep *Sent:* August 16, 2005 12:33 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] quad t1 / 1U rack server combos It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C'mon guys! Certify a few current model servers and be done with it. Without that information I must again ask the question; What 1u server combos work with the new quad pri cards UNDER LOAD (more than 75% channel use). Every user that buys a Digium PRI card should not have to play hit or miss with 2 or 3 servers that cost more than the card to get it to work. Please Please Please publish something useful to support the sale of PRI cards. Damon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Called Party Identification on Polycom IP501
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Tuesday, August 16, 2005 1:21 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Called Party Identification on Polycom IP501 Greetings, The Polycom SIP 1.5 Admin Guide says this: 3.1.8 Connected Party Identification Where possible, the identity of the remote party to which the user has connected is displayed and logged. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion. Does anyone know if * can provide the network signaling required? If so, how? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp That is very dependent on how the call egresses from *, ISDN, POTS, SIP, ??? Who are you calling? If I recall correctly it will work when you call another extension on the * box, but the signaling for that info does not exists in PRI/T1/POTS, so it is not an * issue if you area calling out, * cant get the info from the telco, so * cant send it to the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quad t1 / 1U rack server combos
I have to agree, 4xT1 density is too low for $2500. If there is some magic sauce inside the box then maybe. What exactly is it? A 4 BRI card in a mini Linux install? Who maintains the SIP-ISDN translations? What about docs and support? What are the chances the box is really just an mini * server? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Tuesday, August 16, 2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] quad t1 / 1U rack server combos I think the foneBRIDGE is too expensive for what it does. IMHO -jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Tuesday, August 16, 2005 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] quad t1 / 1U rack server combos William - You should take a look at the foneBRIDGE, new product from redFONE. It has (4) PRI interfaces, and you run our to your primary and failover Asterisk servers via Ethernet. It does not do load balancing. but if you have a hardware failure in your primary Asterisk box, you can just fail right over to your secondary box. You don't need any PRI interface cards in your Asterisk host server at all. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory William Boehlke wrote: In our opinion, BAD idea to put four T1s on a single box, unless you have another box that also has 4 T1s. When, not if, the board fails, you have to take your box down to replace it. And as with anything having to do with computers you are guaranteed a failure at a peak time. Better to split the load between two boxes. William Boehlke Signate *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Chad Osmond *Sent:* Tuesday, August 16, 2005 12:15 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [Asterisk-Users] quad t1 / 1U rack server combos From what I understand (From Sangoma's tech support) and having a IBM x306 SCSI system with an A102u I believe that the system will scale up to 4xT1's easily. With a full T1 of traffic coming in and playing music on hold, the CPU was at 7% with no transcoding. Sangoma cards are supposed to place less draw on the interrupts and offer some new direct writing to DMA in their A104 cards. You may want to give them a call (Scott or Nenad are the two best people to speak with). From Sangoma README.asterisk: * Voice data is channelized and grouped into 8 byte chunks in HARDWARE. Each voice channel is then DMAed directly into the ZAPTEL buffers. Thus there is ZERO copy from HARDWARE to ZAPTEL, resulting in better performance and scalability.* It sounds to me like that would be once advantage over Digiums cards. They also have Hardware PRI functions that are passed directly to libpri. http://sangoma.com/linux/README.asterisk Hope that helps. Chad *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Damon Estep *Sent:* August 16, 2005 12:33 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] quad t1 / 1U rack server combos It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C'mon guys! Certify a few current model servers and be done with it. Without that information I must again ask the question; What 1u server combos work with the new quad pri cards UNDER LOAD (more than 75% channel use). Every user that buys a Digium PRI card should not have to play hit or miss with 2 or 3 servers that cost more than the card to get it to work. Please Please Please publish something useful to support the sale of PRI cards. Damon -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 8/15/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 8/15/2005
[Asterisk-Users] realtime caching
Can anyone shed some light on realtime caching? My desired behavior is that MWI works with realtime voicemail/sip/extensions AND updates to the database take place on the next call to the extensions. Right now I have rtcachefriends=yes, and MWI works, but updates to the database for a cached user seem to still require a reload. It is my understating that removing rtcachefriends will break MWI? Is that true? Is there a best of both worlds approach? MWI and realtime updates to extensions? I have reviewed the info below from the sip.sample.conf, but I must be dense, still dont get it. ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis. ;rtnoupdate=yes ; do not send the update request over realtime. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered when the registration expires ; the friend will vanish from the configuration until requested ; again. If set to an integer, friends expire ; within this number of seconds instead of the ; same as the registration interval ;rtignoreexpire=yes ; when reading a peer from Realtime, if the peer's registration ; has expired based on its registration interval, used the stored ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] realtime caching
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Tuesday, August 16, 2005 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] realtime caching I have reviewed the info below from the sip.sample.conf, but I must be dense, still don't get it. flips on tv to the asterisk televangelist channel Do you find the RealTime comments in sip.conf just a little too confusing? Are you frustrated by the use of double negatives in configuration options? Do not be afraid. You are not alone. Follow the path to enlightenment and visit: http://bugs.digium.com/view.php?id=4075; It is my understating that removing rtcachefriends will break MWI? Is that true? Yes. What exactly are you trying to accomplish? Are your peers/users not being updated in your database? Are you sure? Are you watching debug for SQL log? -Matthew We have a web interface where users can update their dialplan online (not in production yet). The web page modifies the mySQL record. It seems that some options are not re-read when caching is on, for example, changing the caller ID value in the sip table has no effect until a reload (or expiration), so at least in some cases rtcahcefriends makes realtime notsorealtime. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Called Party Identification on Polycom IP501
Try quotes and no spaces between name and number. Callerid=first last2471 -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Tuesday, August 16, 2005 5:31 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Re: Called Party Identification on Polycom IP501 Hi Damon, It's not working SIP to SIP - I am wondering if there is something I am missing in my * config. What I see on the Polycom display is: To:2471 2471 Called party entry in sip.conf (calling party entry is identical): [2471] type=friend context=internal callerid=C* M 2471 secret= host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 [EMAIL PROTECTED] The called party entry in phone2471.cfg (calling party entry is identical): ?xml version=1.0 encoding=UTF-8 standalone=yes? !-- Example Per-phone Configuration File -- !-- $Revision: 1.59 $ $Date: 2004/05/22 00:50:41 $ -- phone2471 reg reg.1.displayName=C* M reg.1.address=2471 reg.1.label=2471 reg.1.type=private reg.1.auth.userId=2471 reg.1.auth.password=/ msg msg.bypassInstantMessage=1 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=*98/ /msg /phone2471 Am I missing anything? Regards, Anthony That is very dependent on how the call egresses from *, ISDN, POTS, SIP, ??? Who are you calling? If I recall correctly it will work when you call another extension on the * box, but the signaling for that info does not exists in PRI/T1/POTS, so it is not an * issue if you area calling out, * cant get the info from the telco, so * cant send it to the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Security and SIP
Block sip on a firewall between * and the public internet, and then create rules for your peers IP range. This assumes you know the IP that all peers and client use; if not just block from regions of the world you do not need to connect to/from. We find that most hack attempts come from one well known region, so we block the entire IP range routed to that region. Also, add noload= for the voip protocols you do not use in modules.conf. You are far better off even if you do things like limiting the connections to the ENTIRE ip range of your local Cable/DSL providers. Prevents folks in the rest of the world from even trying to connect. Toll fraud is huge, it looks like you have done the basics, but you should take additional steps many other would call unnecessary since you will get the bill if someone gets it. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of John Fawcett Sent: Monday, August 15, 2005 3:22 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Security and SIP I've now setup SIP for: - internal softphones - registering with external providers (like FWD) for making calls - receiving calls from theese providers For the latter step, it was necessary to forward ports from my NAT to the asterisk server: 5060 + range of ports mentioned in rtp.conf. I was just wondering about how to make this setup as secure as possible. Here's what I've done so far: 1. defined a default context in sip.conf which cannot access any real extension. sip.conf: [general] context=from-unknown-sip extensions.conf: [from-unknown-sip] exten = _.,1,CONGESTION 2. for peers, defined a context which does not provide access to outside lines. sip.conf: [fwd.pulver.com] type=peer username=688426 fromuser=688426 secret=xx host=fwd.pulver.com port=5060 nat=yes canreinvite=no insecure=very context=sip-external disallow=all allow=ulaw 3. for peers, defined insecure=very which should check that the incoming call comes from the same IP as was registered. 4. for internal softphones, which can make outgoing calls, limited registrations to a specific network address using deny/permit sip.conf: [31] type=friend callerid=[EMAIL PROTECTED] 31 host=dynamic deny=0.0.0.0/0.0.0.0 permit=192.168.2.32/255.255.255.255 context=sip-internal secret= disallow=all allow=ulaw allow=alaw Anything else I can do to improve security? I specifically don't want anyone external to be able to make calls. As I've opened port 5060 + rtp.conf ports only for the purpose of receiving calls from services I have registered with, I don't want any external phones to be able to register via this route. Is there any risk of this if someone can guess a password (maybe unlikely but given time this could happen). Thanks, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] premature call release - SIP 480
Damon Estep wrote: When executing: Dial (SIP/[EMAIL PROTECTED],60 mailto:SIP/[EMAIL PROTECTED],60) I get about 15 seconds of ringing, the called party rings, but if not answered in the ~15 seconds I get back SIP 480 temporarily unavailable. If the call is answered everything is fine and the call will continue as expected. The call is being passed to a TNT media gateway then to the PSTN via a PRI The TNT reports Q850 cause 19 and responds with SIP 480 Somehow the TNT thinks the called stopped progressing on the PRI after 15 to 20 seconds. The Telco says they have done a capture and are getting a normal release, in other words their switch is not terminating the call or sending any Q850 message. I can not find any timers in the TNT that might cause this, and it is not reporting any expired timers. Any ideas? Does the SIP INVITE from * to the TNT contain a timeout? If so is it possible the, 60 in the dial command is being ignored? Either; The TNT got a maximum time parameter from asterisk and it has been exceeded, so the TNT responds 480, or; The TNT has a timer that expires after n seconds and sends the 480 on its own, or; The Telco is not seeing the progress they want to see and is sending the Q850 cause 19. Any opinions, suggestions? Do you have qualify= on ? This ended up being a global dial out timer on the media gateway (MAX TNT) in the SYSTEM profile. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail - 99 message limit
Thanks Luki, Seems easy enough, does the code look like it would be hard to change that value from a hard coded value to a global variable which can be defined in voicemail.conf and overridden for a single mailbox? I am not a coder so an opinion would be useful. I have cross posted this to -dev since it seems to be going that route. Damon See apps/app_voicemail.c: #define MAXMSG 100 Then recompile the app and reload the module (or restart asterisk). --Luki On 8/12/05, Damon Estep [EMAIL PROTECTED] wrote: Anyone know how to override the 99 message limit in voicemail? (yeah, we have a public VM that gets that many a day). ___ ___ 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] premature call release - SIP 480
When executing: Dial (SIP/[EMAIL PROTECTED],60) I get about 15 seconds of ringing, the called party rings, but if not answered in the ~15 seconds I get back SIP 480 temporarily unavailable. If the call is answered everything is fine and the call will continue as expected. The call is being passed to a TNT media gateway then to the PSTN via a PRI The TNT reports Q850 cause 19 and responds with SIP 480 Somehow the TNT thinks the called stopped progressing on the PRI after 15 to 20 seconds. The Telco says they have done a capture and are getting a normal release, in other words their switch is not terminating the call or sending any Q850 message. I can not find any timers in the TNT that might cause this, and it is not reporting any expired timers. Any ideas? Does the SIP INVITE from * to the TNT contain a timeout? If so is it possible the, 60 in the dial command is being ignored? Either; The TNT got a maximum time parameter from asterisk and it has been exceeded, so the TNT responds 480, or; The TNT has a timer that expires after n seconds and sends the 480 on its own, or; The Telco is not seeing the progress they want to see and is sending the Q850 cause 19. Any opinions, suggestions? ___ 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] voicemail - 99 message limit
Anyone know how to override the 99 message limit in voicemail? (yeah, we have a public VM that gets that many a day). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Blank CIDName or CIDNum = asterisk
So caller ID name is passed when available and nothing is passed when not? That worked. The following line also got rid of asterisk without entering any custom info: callerid= Thank you, Hugh On Thu, 2005-08-11 at 03:19 +0100, Tony Hoyle wrote: In the [default] section of sip.conf put: callerid=unavailable ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime + MYSQL
Rollin, My real-time works fine, Nathan was the original poster o this message. I simply added the table structure for real-time voicemail J Damon, You may be querying the wrong table, because the following fields in your Select statement do not exit in the table, voicemail_users, that you created: category, var_name, var_val, cat_metric, filename, commented Every item mentioned in a Select query must exist in the table that is being queried. Rollin Weeks On 8/10/05, Damon Estep [EMAIL PROTECTED] wrote: I'm having a few issues with the MySQL realtime configuration in CVS-HEAD. I tested it initially with realtime extensions (realtime_ext = mysql,asterisk,extensions) and a realtime switch in extensions.conf and that works fine, So I though I'd go back and test a static configuration mapping. I used the table structure from the asterisk guru postgres howto to create something similar in MySQL (shown below) and included the following in extconfig; voicemail.conf = mysql,asterisk,voicemail_users The result is that app_voicemail fails to load and it appears from the debug that it is not happy with the table structure... however the names it has for the fields seem strange (to me that is :)) If anyone has gone through the process of creating the correct tables in MySQL and doesn't mind sharing I would be most appreciative. Regards, Nathan. MySQL Table CREATE TABLE voicemail_users ( id int NOT NULL auto_increment, customer_id varchar(255) NOT NULL default '0', context varchar(255) NOT NULL default '', mailbox varchar(255) NOT NULL default '', password varchar(4) NOT NULL default '0', fullname varchar(50) NOT NULL default '', email varchar(50) NOT NULL default '', pager varchar(50) NOT NULL default '', stamp datetime NOT NULL default '-00-00 00:00:00', PRIMARY KEY(`id`) ); ### res_mysql.conf [general] dbhost = localhost dbname = asterisk dbuser = asterisk dbpass = dbport = 3306 dbsock = /var/run/mysqld/mysqld.sock Debug Log Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Static SQL: SELECT category, var_name, var_val, cat_metric FROM voicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Everything is fine. Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Query: SELECT category, var_name, var_val, cat_metric FROM voicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Query Failed because: Unknown column 'category' in 'field list' ___ This works for voicemail in CVS-HEAD CREATE TABLE `voicemail` ( `uniqueid` int(11) NOT NULL auto_increment, `customer_id` int(11) NOT NULL default '0', `context` varchar(50) NOT NULL default '', `mailbox` varchar(10) NOT NULL default '0', `password` varchar(4) NOT NULL default '0', `fullname` varchar(50) NOT NULL default '', `email` varchar(50) NOT NULL default '', `pager` varchar(50) NOT NULL default '', `stamp` timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP, PRIMARY KEY(`uniqueid`), KEY `mailbox_context` (`mailbox`,`context`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime + MYSQL
I'm having a few issues with the MySQL realtime configuration in CVS-HEAD. I tested it initially with realtime extensions (realtime_ext = mysql,asterisk,extensions) and a realtime switch in extensions.conf and that works fine, So I though I'd go back and test a static configuration mapping. I used the table structure from the asterisk guru postgres howto to create something similar in MySQL (shown below) and included the following in extconfig; voicemail.conf = mysql,asterisk,voicemail_users The result is that app_voicemail fails to load and it appears from the debug that it is not happy with the table structure... however the names it has for the fields seem strange (to me that is :)) If anyone has gone through the process of creating the correct tables in MySQL and doesn't mind sharing I would be most appreciative. Regards, Nathan. MySQL Table CREATE TABLE voicemail_users ( id int NOT NULL auto_increment, customer_id varchar(255) NOT NULL default '0', context varchar(255) NOT NULL default '', mailbox varchar(255) NOT NULL default '', password varchar(4) NOT NULL default '0', fullname varchar(50) NOT NULL default '', email varchar(50) NOT NULL default '', pager varchar(50) NOT NULL default '', stamp datetime NOT NULL default '-00-00 00:00:00', PRIMARY KEY (`id`) ); ### res_mysql.conf [general] dbhost = localhost dbname = asterisk dbuser = asterisk dbpass = dbport = 3306 dbsock = /var/run/mysqld/mysqld.sock Debug Log Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Static SQL: SELECT category, var_name, var_val, cat_metric FROM voicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Everything is fine. Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Query: SELECT category, var_name, var_val, cat_metric FROM voicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Query Failed because: Unknown column 'category' in 'field list' ___ This works for voicemail in CVS-HEAD CREATE TABLE `voicemail` ( `uniqueid` int(11) NOT NULL auto_increment, `customer_id` int(11) NOT NULL default '0', `context` varchar(50) NOT NULL default '', `mailbox` varchar(10) NOT NULL default '0', `password` varchar(4) NOT NULL default '0', `fullname` varchar(50) NOT NULL default '', `email` varchar(50) NOT NULL default '', `pager` varchar(50) NOT NULL default '', `stamp` timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP, PRIMARY KEY (`uniqueid`), KEY `mailbox_context` (`mailbox`,`context`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1; ___ 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] inbound caller id name pri - tnt - asterisk
Anyone out there have success getting caller id name from a pri, through a lucent tnt, to asterisk? What about from other media gateways? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN DID
How many digits is your pri provider sending in the setup message? It needs to match your dilaplan, ie if they are sending 4 you need 4 digit extensions or some other monkey business to translate. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Panitaxx Sent: Tuesday, August 09, 2005 4:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ISDN DID Hello, I have an ISDN PRI E1. For some reason I am not receiving the did number so every call can only go to s exten. I have tried using _X. exten. Also I have immediate=no in zapata.conf. Any hint? thanks in advance, Iván Aponte ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] include behavior (word puzzle of the day)
The key seems to be listing the 10 digit extensions dialplan in a context other than the context they are defined in in sip.conf, correct? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dbruce Sent: Thursday, August 04, 2005 6:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] include behavior (word puzzle of the day) Try something like this: [context1] Include = internal-extensions include = egress [context2] include = egress [context3] include = pri-ingress include = internal-extensions [internal-extensions] ;sip users with 10 digit extensions [egress] ;media gateway terminating local 10 digit calls [pri-ingress] ;inbound PRI via media gateway Regards, Derek - Original Message - From: Damon Estep To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, August 04, 2005 6:26 PM Subject: [Asterisk-Users] include behavior (word puzzle of the day) In the example below context2 is included in context3 because it is included in context1. Is there a way to include context2 in context1, and context1 in context3, but not context2 in context3 as a result. [Context1] ;sip users with 10 digit extensions Include = context2 [context2] ;media gateway terminating local 10 digit calls [context3] ;inbound PRI via media gateway Include = context1 I have a case where a dialplan is insecure because inbound calls in context3 can be re-routed back out in context2. Actually, what occurs is a loop, where the call comes in context3, finds no match in context1, egresses in context2, and repeats the loop, setting up a lot of calls in a short period of time! Extensions in context1 need to be able to reach extensions in context2 Inbound calls into context3 need to be able to reach extensions in context1 Inbound calls in context3 MUST be restricted from reaching extensions in context2 which are outside extensions sent out to a SIP provider. It would seem more logical and secure if includes did not cascade, or would not make 2 hops Perhaps I have failed to understand some simple concept that would resolve this issue? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] include behavior (word puzzle of the day)
In the example below context2 is included in context3 because it is included in context1. Is there a way to include context2 in context1, and context1 in context3, but not context2 in context3 as a result. [Context1] ;sip users with 10 digit extensions Include = context2 [context2] ;media gateway terminating local 10 digit calls [context3] ;inbound PRI via media gateway Include = context1 I have a case where a dialplan is insecure because inbound calls in context3 can be re-routed back out in context2. Actually, what occurs is a loop, where the call comes in context3, finds no match in context1, egresses in context2, and repeats the loop, setting up a lot of calls in a short period of time! Extensions in context1 need to be able to reach extensions in context2 Inbound calls into context3 need to be able to reach extensions in context1 Inbound calls in context3 MUST be restricted from reaching extensions in context2 which are outside extensions sent out to a SIP provider. It would seem more logical and secure if includes did not cascade, or would not make 2 hops Perhaps I have failed to understand some simple concept that would resolve this issue? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cvs Head
We recently upgraded a production system to current cvs head, things are working well. We do use queues extensively. There were two bugs in our environment that have been fixed as of 8/3/2005, one was a segfault in voicemail if a user did not enter a password and hung up, the other was the failure to recognize the * key in a macro with a a priority. I suggest you back up what you have now and test the most recent cvs head, unless of course the most recent stable release has the features you need in which case you should use it. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jennifer Hales Sent: Thursday, August 04, 2005 6:24 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cvs Head Hello Asterisk Users, Does anyone have a stable cvs head release date you can recommend? It will need to be deemed stable in a queue environment. We are currently running Centos 2.6 kernal and have implemented different versions of cvs head with varying results. I am currenly using cvs head 20/05/2005 however it is not utilizing the wrapuptime function in Agents.conf. We are real close as this appears to be our only problem. Appreciate your help Kind regards Jennifer Hales ___ 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] priority a in macro to access voicemail
I have added the following to a macro that is used for all
extensions so a user can access voicemailmain by pressing * during the
voicemail prompt
; check voicemail
exten = a,1,voicemailmain(${macro_exten})
exten = a,2,hangup
The behavior is a little weird, the * key is not recognized
during the portion of the greeting where the extension number is being played
back, after it is played back, for the duration of the greeting, the * key is
recognized and works as expected.
Any ideas?
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RE: [Asterisk-Users] priority a in macro to access voicemail
I missed the part of that page that has anything to do with the
question. The portion of the dialplan I posted is a small snipet of a
huge macro, that part that sends you to voicemailmain when * is pressed.
It works, but has a small bug as previously stated;
The behavior is a little weird, the * key is not recognized during the
portion of the greeting where the extension number is being played back,
after it is played back, for the duration of the greeting, the * key is
recognized and works as expected.
i think may be you should read this:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Macro
On 8/2/05, Damon Estep [EMAIL PROTECTED] wrote:
I have added the following to a macro that is used for all
extensions so
a
user can access voicemailmain by pressing * during the voicemail
prompt
; check voicemail
exten = a,1,voicemailmain(${macro_exten})
exten = a,2,hangup
The behavior is a little weird, the * key is not recognized during
the
portion of the greeting where the extension number is being played
back,
after it is played back, for the duration of the greeting, the * key
is
recognized and works as expected.
Any ideas?
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RE: [Asterisk-Users] * behind NAT and local subnet
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Friday, July 15, 2005 1:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * behind NAT and local subnet Asterisk shows failed to authenticate user. This is clearly NAT related as the same user works fine inside the NAT with no config changes What phone? How is the server and proxy info configured? There is no problem witht he setup assuming ports are set up properly. Sounds more like a wrong entry in the phone for outgoing proxy, or something. Snom 190 phone Registrar is set to the public IP address side of the NAT in front of the * box. Under advanced line settings outbound proxy is null Phone registers, and can receive calls. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * behind NAT and local subnet
Thanks for your help, I think setting the externip breaks non NATd clients because it mangles the SIP headers by spoofing the source IP of SIP messages, correct? I was able to resolve this issue by upgrading the Cisco router IOS to 12.3-15 which apparently does a better job with SIP header NAT. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julian J. M. Sent: Friday, July 15, 2005 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * behind NAT and local subnet Have you set correctly the externip and localnet keywords in sip.conf? Julian. On 7/15/05, Damon Estep [EMAIL PROTECTED] wrote: I have an * box behind a NAT router (static NAT, port ACLs set up correctly) Most of the SIP users are on the local subnet with the * box, they work fine Take one of the same users off of the local subnet and come in through the NAT router and these results; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] arrgg! www.voip-info.org down again (or too busy)
Does anyone have a mirror of this running? ___ 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] * behind NAT and local subnet
I have an * box behind a NAT router (static NAT, port ACLs set up correctly) Most of the SIP users are on the local subnet with the * box, they work fine Take one of the same users off of the local subnet and come in through the NAT router and these results; The remote user can register The remote user can receive calls The remote user can get into voicemail not sure why The remote user can not place calls The trace shows 407, proxy authentication required Asterisk shows failed to authenticate user. This is clearly NAT related as the same user works fine inside the NAT with no config changes Is there are a way to connect to an Asterisk server that sits behind NAT without breaking the ability to connect form the local non-NATed subnet? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on Linksys WRT54G
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Walid Azab
Sent: Tuesday, July 05, 2005 4:23
AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk
on Linksys WRT54G
Hi all,
Any one tried installing Asterisk on Linksys WRT54G? We have
but facing problems with SIP to SIP calls. The phones ring and calls are
established but we cannot hear any voice at all. I tried allow=all in the
general section but did not work. So I forced ulaw. Can any one please check it
out and let me know what is wrong?
Here are the conf files:
Asterisk Version: Asterisk
CVS-HEAD-01/17/05-00:35:58 built by [EMAIL PROTECTED]
on a i686 running Linux
==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 all codecs
allow=ulaw
context = bogon-calls ; Send SIP callers that we don't know about here
[2000]
type=friend
; This device takes and makes calls
username=2000 ; Username on
device
secret=1234 ;
Password for device
host=dynamic ; This host
is not on the same IP addr every time
context=from-sip ; Inbound calls from this host
go here
mailbox=100 ;
Activate the message waiting light if this
; voicemailbox has messages in it
[2001]
; Duplicate of 2000, except with different auth data
type=friend
username=2001
secret=1234
host=dynamic
context=from-sip
mailbox=101
==Extensions.conf
[general]
static=yes
writeprotect=yes
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,2,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
exten = 2001,1,Dial(SIP/2001,20)
exten = 2001,2,Voicemail(u2001)
exten = 2001,102,Voicemail(b2001)
exten = 2001,103,Hangup
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
How are the routers connected to the IP
network? Any nat before them on either end?
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RE: [Asterisk-Users] How does Vonage support fax machines?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Deon Sent: Tuesday, July 05, 2005 8:32 AM To: Asterisk Users Subject: [Asterisk-Users] How does Vonage support fax machines? My boss is insisting we support fax, and I keep telling him that Fax over IP is very unreliable and not recommended and his immediate come-back is Vonage does it. and it's very hard to figure out how. I don't think Vonage does T.38, the Linksys/Sipura units they're using doesn't support T.38 to my knowledge. That means they have to be using G.711Ulaw to send faxes. But how do they compensate for packet loss/jitter/etc. In our test lab, the best we could get was 90% success at sending faxes. It seemes to screw up the longer the transmission, ie page 1 was usually ok, but page 2 and 3 and 4 was at serious risk. So if I bought a Vonage adapter, can I send a 30 page fax? My best guess is they have high quality voice T1's, like from an ILEC, usually more expensive, and when they sell a Fax Line I noticed it's more expensive. Maybe they route all their fax calls specifically out these high quality T1's that they own, so that they can do some type of quality control. My test lab was a private network, a Cisco 3640 connected to a local voice PRI T1, and converting to SIP. Asterisk would push the calls to the Cisco 3640 and the Linksys PAP2 would register with Asterisk. All local. I then tried several test faxes throughout the PSTN. Would it be better to plug the voice T1 straight into Asterisk using one of Digium's cards? We relay faxes to/from a PRI connected to * to a fax machine using a sipura spa-2000 and g.711u. The span between * and the spa-2000 is a single LAN switch, 100mbit. While t.38 would be better, it appears to us that g.711u works well on low latency links, so supporting fax over IP would likely require QoS implementation on the IP link if there are latency/jitter issues. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How does Vonage support fax machines?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Tuesday, July 05, 2005 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How does Vonage support fax machines? What makes you think they do? The marketing pieces? We all know G71 is not reliable for faxing, and for Vonage to advertise it is irresponsible of them. mailman/listinfo/asterisk-users They clearly advertise it, and the even indicate that you use port 2 on your ATA! They support it the same way they support QoS, if it does not work they tell you to call your ISP. Here is the link http://www.vonage.com/features.php?feature=fax ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [Fwd: Asterisk Balancing solution]
Dear All, I am using Linux-High Availability between two Asterisk servers, everything is fine but I do have one problem with this, When a server fails and the other assumes the ip address and start asterisk on server 2, the ip phone must re-register themselves again, otherwise the phones are dead. Does anyone have Ideas of how to overcome this. I would love to see someone get this to work! My thoughts in the past are to do a periodic sip show peers and save the data to the standby server in a text file or to a mysql database.The question then becomes how to re-register the peer manually, which I am sure there is an answer for , but I do not have it.If you ever get it working you should post!Other things to consider voicemail files, MWI status, etc. have you been able to address these issues? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with OR Logic in the GotoIf Statement
If you need a fast solution put two gotoif
statements in a row, one to check for the first condition, another to check for
the next, you can leave out the redirect If the condition is not matched so it
just goes to the next priority.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Keith O'Brien
Sent: Wednesday, June 29, 2005
8:40 AM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Problems
with OR Logic in the GotoIf Statement
I am having some trouble implementing OR login in the
GotoIf statement. I have followed the examples in the Wiki and I
still am getting a syntax error.
Essentially I want to screen for CallerIDs set for
Anonymous OR Unknown Caller. If either of
these are true I want to send it to statement 3 which clears the CallerID and
proceeds to Privacy Manager.
I have also tried removing and adding quotes to no
avail. I am running the 6/7/2005 CVS Head.
exten =5000,1,NoOp,${CALLERIDNAME}
exten =5000,2,GotoIf($[$[${CALLERIDNAME} =
Anonymous] | $[${CALLERIDNAME} = Unknown
Caller]]?3:5)
exten =5000,3,SetCIDNum()
exten =5000,4,SetCIDName()
exten =5000,5,PrivacyManager
exten =5000,6,GotoIfTime(19:00-7:00|*|*|*?afterhours,s,1)
exten =5000,7,agi,astcallerid
exten =5000,8,DIAL(SIP/5001)
exten =5000,9,Voicemail(u5001)
exten =5000,110,Hangup
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-2, Anonymous) in new stack
Jun 29 10:34:09 WARNING[3946]: ast_expr.y:486 ast_yyerror: ast_yyerror():
syntax error: syntax error; Input:
(Anonymous = Anonymous)|(Anonymous = Unknown Caller)
^^^
^
-- Executing GotoIf(IAX2/[EMAIL PROTECTED]:4569-2, 0?3:5) in new stack
-- Goto (in-out,7326031000,5)
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RE: [Asterisk-Users] simultaneus calls?
The 1 m internet connection will be the limiting factor in your setup, you did not state what type of internet connection, but given the speed of 1 mbit it must be DSL (or maybe fraction t/e1). Is the outbound speed also 1m? Is there data on the line also? How much? What about voice Qos? You should start here http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erdem HAKI Sent: Tuesday, June 28, 2005 3:04 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] simultaneus calls? Hello, How can i learn my asterisk how many simulyaneus calls support? My configuration: 80 GB HDD, 1 GB Ram, P4 2,8 MHz processor, Fedora Core 3 minimum installation, no digium cards, codecs g729 or gsm, 1Mbit internet connection. Thanks for your interest... Erdem HAKI [EMAIL PROTECTED] ___ 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] Disable record busy greeting option in voicemail
I have an application that calls for a single greeting to be used exclusively in a voicemail box (rather than busy/unavailable). It is simple enough to implement in the dialplan, but is there a way to remove the option in the voicemail menu to record the busy greeting which only serves to confuse users in this scenario? Look and could not find my answer in the usual spots. Thx! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial peer preference
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of kurt x Sent: Friday, June 24, 2005 6:24 AM To: Asterisk Subject: [Asterisk-Users] Dial peer preference Does Asterisk support preference for the dial peers. For example: I have two outbound peers in *. The first is a SIP dial peer and the second peer is to the PSTN via a T1. The SIP dial peer is the main outbound peer for all calls. However, if the my SIP providers network goes down, I need to be able to automatically route the call out the T1 card. Is this possible in Asterisk. I have not seen any preference commands for Asterisk. If not, is there a work around for this type of set up. Kurt Have you tried putting in something like this? Etxen s,1,Dial(sip/[EMAIL PROTECTED],duration) Exten s,2,Dial(zap/chan/number,duration) Exten s,3,Congestion(5) Exten s,4,(hangup) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] logged in agent make an outbound call?
Yes, I know. In this case the agent is logging in from a remote phone (pots line) and staying logged in. If they used agentcallbacklogin they could make outbound calls, but the long distance bill would hit their line, not the * box... You could use Agentcallbacklogin instead - the queue will call them when a call comes in, but they are free to make outbound calls in the meantime. Julian. Damon Estep wrote: Is there a way for a logged in agent (hearing music on hold) to initiate an outbound call without logging out of the queue? We want sales agents to be able to make outcalls when there is no callers in queue, but still be logged in to get new inbound calls if they come in. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip-info.org unreliable lately?
Damon Estep wrote: I assume the bandwidth is being donated or something, but surely someone would be willing to donate reliable bandwidth as the knowledge hosted on the site (which is also donated!) is worth way more than the bandwidth. Sure it's the bandwidth? If the wiki is loaded, I see Server load on the bottom of the page, the numbers sometimes go as high as 80-100..? Not sure if it's a Linux (guess so? :p) but if that represents the system load.. 80 is a 'bit' high indeed. Cheers 80-100 might be a lot for the current environment, but given the number of * users it is very small. Point is the server and bandwidth should be able to handle a lot more users if we are all going to rely on it as the (un)official repository for * guides. I have seen many posts from users willing to pitch in, but still have no idea where the site is now or what the arrangement is. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] logged in agent make an outbound call?
Calls into the asterisk box, including non-VoIP remote agents, are via a ISDN/PRI on a Digium T1 card. It is the same PRI that inbound and outbound calls come in on and go out through, there are no IP dial tone providers. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 22, 2005 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] logged in agent make an outbound call? What type of connection are you using to link their pots with * ? For the inbound part, * would be calling them to connect the call. For their outbound, could you not use the same mechanism that you are currently using to login, but dial the outbound number instead (so it is * doing the dialling) ? I would have thought that if they can call * to login, then they can call * to make an outbound call . Julian. Damon Estep wrote: Yes, I know. In this case the agent is logging in from a remote phone (pots line) and staying logged in. If they used agentcallbacklogin they could make outbound calls, but the long distance bill would hit their line, not the * box... You could use Agentcallbacklogin instead - the queue will call them when a call comes in, but they are free to make outbound calls in the meantime. Julian. Damon Estep wrote: Is there a way for a logged in agent (hearing music on hold) to initiate an outbound call without logging out of the queue? We want sales agents to be able to make outcalls when there is no callers in queue, but still be logged in to get new inbound calls if they come in. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voip-info.org unreliable lately?
Anyone have any insight as to why voip-info.org has been up and down all day, and more importantly unreliable for the last month? I assume the bandwidth is being donated or something, but surely someone would be willing to donate reliable bandwidth as the knowledge hosted on the site (which is also donated!) is worth way more than the bandwidth. There is no doubt it is the best documentation that exists on *, but only when accessible. Gripe, gripe, gripe ___ 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] logged in agent make an outbound call?
Is there a way for a logged in agent (hearing music on hold) to initiate an outbound call without logging out of the queue? We want sales agents to be able to make outcalls when there is no callers in queue, but still be logged in to get new inbound calls if they come in. ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Listen to multiple ports
If you do a sip show peers I think you will see that your PAP2 setup registers its port with * as being 5060 on line 1 and 5061 on line 2, but it stills calls port 5060 on asterisk when it makes the registration. I think * is actually listening on the first configured port. You might get the same results you have now after removing the port=5061, have you tried Prepaid, post if this actually works or not in your case where port 5060 out from the UA is blocked. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, June 13, 2005 11:41 PM To: Prepaid; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Listen to multiple ports Hello I added to sip.conf 2 instances of the port parameter ej: [general] port=5060 port=5061 It works for me, we used the Linksys pap2-na with both lines at the same time we cant bind it to the same 5060 port, then I configured line1 to 5060 and line2 to 5061, asterisk is listening at both ports and working like expected Juan Bou At 12:09 a.m. 14/06/2005, you wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Prepaid Sent: Monday, June 13, 2005 10:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP Listen to multiple ports Hello all I'm trying to get my asterisk config to listen to multiple ports. This is since some clients have port 5060 blocked by their ISP. Does anyone know how to do this in sip.conf or if it is even supported? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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 ~ - Juan Bou Riquer. Internet Cancun. [EMAIL PROTECTED] Tel. 87-2601 Fax. 84-3809 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB
Forget about MS SQL, odbc drivers that run on linux to talk to MS SQL stink Odbc in general stinks. You might be able to get MS SQL DTS (data transformation services) to link to the mysql database and present the data as it were in your ms sql database. There is a pretty good odbc 3.51 mysql driver for windows, as well as a .net provider. Both at www.mysql.org. Mysql is free, * will talk to is using the native TDS You can run the windows version of mysql on a windows box if you wish, but why? Faster if it is on the same box as asterisk unless * is heavily loaded. I tried the * realtime odbc mssql thing, gave up after having poor results getting the various ms sql drivers for linux to work right. our main app uses data in ms sql and mysql and there is a common key in the data to link accounting data with the * user data for views where they are both required. We also use mysql for cdr for billing purposes. I was much more comfortable with .net ms sql, but the transition and integration with mysql was easy. Just store the asterisk specific data in mysql, everthing else in ms sql if you must. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili Sent: Tuesday, June 14, 2005 12:04 AM To: 'Shamsul Arefin'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB Could you go with some details? What was better performance, stability? All our user info is in MS SQL and we have billing based on it, so it won't be easy to move to mysql. I.N. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shamsul Arefin Sent: Monday, June 13, 2005 10:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB Yes it is now possible to store configuration files in database, via Mysql support or via ODBC. But we have find that Mysql is works much better. regards shams On 6/14/05, Irakli Natsvlishvili [EMAIL PROTECTED] wrote: Hello, I have one question regarding *. Default configuration for asterisk is to keep configuration(s) in ordinary text based config files. My question is simple: is it possible to keep those config info (at least, to start from - sip.conf, extensions.conf and voicemail.conf) on a database, which asterisk access via ODBC. If it is possible, I'd appreciate if someone points me where I can read more about it and shows me some examples. Also I'd like to know, how asterisk behaves (in terms of stability and performance) in this environment. I.N. ___ 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 -- Best Regards Shamsul Arefin Saktek Technologies ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Listen to multiple ports
I just ran a couple of test with CVS Head Port=5060 Port=5061 Result = chan_sip reports listening on 5060 Port=5061 Port=5060 Result = chan_sip reports listening on 5060 (ignoring port=?) Port=5061 Result = chan_sip STILL reports listening on 5060 Bindport=5061 Result = chan_sip reports listening on 5061 Bindport=5061 Bindport=5060 Result = istening on 5060 Conclusion - asterisk only listens on one port, and ignores the second port= or bindport= -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, June 13, 2005 11:41 PM To: Prepaid; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Listen to multiple ports Hello I added to sip.conf 2 instances of the port parameter ej: [general] port=5060 port=5061 It works for me, we used the Linksys pap2-na with both lines at the same time we cant bind it to the same 5060 port, then I configured line1 to 5060 and line2 to 5061, asterisk is listening at both ports and working like expected Juan Bou At 12:09 a.m. 14/06/2005, you wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Prepaid Sent: Monday, June 13, 2005 10:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP Listen to multiple ports Hello all I'm trying to get my asterisk config to listen to multiple ports. This is since some clients have port 5060 blocked by their ISP. Does anyone know how to do this in sip.conf or if it is even supported? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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 ~ - Juan Bou Riquer. Internet Cancun. [EMAIL PROTECTED] Tel. 87-2601 Fax. 84-3809 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB
If your app is .net get the .net provider for mysql and give it to your dba/programmer with the docs, he/she will figure it out. No different than talking to ms sql with .net except you reference a different data provider. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili Sent: Tuesday, June 14, 2005 12:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB Thanks for info . How do you integrate * specific data in mysql with data from MSSQL? App is running on .NET, in this case it will need to have assess to both DBs and update them simultaneously. Sorry, I'm not a DB admin. I.N. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, June 13, 2005 11:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB Forget about MS SQL, odbc drivers that run on linux to talk to MS SQL stink Odbc in general stinks. You might be able to get MS SQL DTS (data transformation services) to link to the mysql database and present the data as it were in your ms sql database. There is a pretty good odbc 3.51 mysql driver for windows, as well as a .net provider. Both at www.mysql.org. Mysql is free, * will talk to is using the native TDS You can run the windows version of mysql on a windows box if you wish, but why? Faster if it is on the same box as asterisk unless * is heavily loaded. I tried the * realtime odbc mssql thing, gave up after having poor results getting the various ms sql drivers for linux to work right. our main app uses data in ms sql and mysql and there is a common key in the data to link accounting data with the * user data for views where they are both required. We also use mysql for cdr for billing purposes. I was much more comfortable with .net ms sql, but the transition and integration with mysql was easy. Just store the asterisk specific data in mysql, everthing else in ms sql if you must. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili Sent: Tuesday, June 14, 2005 12:04 AM To: 'Shamsul Arefin'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB Could you go with some details? What was better performance, stability? All our user info is in MS SQL and we have billing based on it, so it won't be easy to move to mysql. I.N. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shamsul Arefin Sent: Monday, June 13, 2005 10:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB Yes it is now possible to store configuration files in database, via Mysql support or via ODBC. But we have find that Mysql is works much better. regards shams On 6/14/05, Irakli Natsvlishvili [EMAIL PROTECTED] wrote: Hello, I have one question regarding *. Default configuration for asterisk is to keep configuration(s) in ordinary text based config files. My question is simple: is it possible to keep those config info (at least, to start from - sip.conf, extensions.conf and voicemail.conf) on a database, which asterisk access via ODBC. If it is possible, I'd appreciate if someone points me where I can read more about it and shows me some examples. Also I'd like to know, how asterisk behaves (in terms of stability and performance) in this environment. I.N. ___ 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 -- Best Regards Shamsul Arefin Saktek Technologies ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
RE: [Asterisk-Users] SIP Listen to multiple ports
You must have missed the part where Prepaid got upset when I suggested workarounds :) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Juan J. Sierralta P. Sent: Tuesday, June 14, 2005 12:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Listen to multiple ports Hi, What about using IPTABLES DNAT stuff in order to map all incoming 5061 traffic to 5060 port ? That may work. On 6/14/05, Damon Estep [EMAIL PROTECTED] wrote: Conclusion - asterisk only listens on one port, and ignores the second port= or bindport= -- Juanjo sin .sig :( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VOIP-INFO down?
Second day in a row... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Marcel van Kaam, Fonetica Sent: Tuesday, June 14, 2005 8:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VOIP-INFO down? Hi all, Is VOIP-info down? Marcel van Kaam Fonetica Teleservices ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Authentication
Sounds like to much use of the general context, remove etensions from general that you require authentication for or use includes. Post you extensions.conf for better help. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stojan Sljivic - GDS Sent: Tuesday, June 14, 2005 1:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Authentication Hi Olle, Do you have any idea why is Asterisk behaving like this? Did you tested this with your Asterisk? Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Tuesday, June 14, 2005 9:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Authentication Stojan Sljivic - GDS wrote: Hi, I have set autocreatepeer=no and it behaves just the same. It seems that the default value is no, or Asterisk does not understand this property. In which version of Asterisk was this property introduced? I use 1.0.5. autocreatepeer is off by default and should not be part of your problem. /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VOIP-INFO down?
What is the deal with voip-info.org, is it a commercial agreement or a donation that has worn out its welcome? Needs more bandwidth or a faster (load balanced) server! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, June 14, 2005 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] VOIP-INFO down? Second day in a row... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Marcel van Kaam, Fonetica Sent: Tuesday, June 14, 2005 8:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VOIP-INFO down? Hi all, Is VOIP-info down? Marcel van Kaam Fonetica Teleservices ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Wow! I never learn so much! Thanks Guys So if I understand correctly, a full T1 should be 1.5Mbps full duplex. And it should support 22 SIP Users at once - Right? Bart Probably closer to 20 depending on setup/teardown frequency. This is only if the line is dedicated VoIP, no other data traffic. Assuming 64k RTP like g.711 You have to decide how much data and how much voip and define rules on your router (traffic shaping or priority queuing, etc.) to enforce QoS. ___ 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] Macro support in realtime
Is there any way to accomplish the following? (searched and searched and can not find any examples) In extensions.conf (text file) define a macro that accepts a handful of arguments From realtime mysql (extensions) - call the macro with arguments (where the macro is static in the text file) If not, what about putting the macro in mysql? Just trying to find a way to reduce the number of db records per extension to 1 from 6+ by calling a macro with 6+ arguments from a single record. Possible? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM, Asterisk and Attended transfer (bug?)
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Davies Sent: Monday, June 13, 2005 6:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SNOM, Asterisk and Attended transfer (bug?) Hi, I am using a number of snom190 phones, and an asterisk gateway server, and recently started experimenting with call transfers. The snom phones provide support for attended and un-attended call transfer, so I would rather use that than call-parking. I have found that un-attended transfer works fine, and that attended transfer works fine if the originating phone call is NON-SIP (ie. ISDN) I hope that some of this makes sense... When I look at the SIP trace for the sequence of A calls B and is transferred to C, I see: A makes call to B: A calls B B picks up A and B are bridged (re-INVITEd) and talk directly. B then puts A on hold: (A and B are both INVITE to talk via Asterisk) B makes a call to C, I see: B calls C C picks up B and C are bridged (re-INVITEd) and talk directly. B presses transfer: (Same as putting B and C on hold, B and C are re-INVITEd to talk via Asterisk) B selects which line to transfer to C B REFERs A to C by asking Asterisk. Asterisk accepts this. B is notified that A is disconnected B gets BYE for call to A B gets BYE for call to C C gets INVITE to talk to B via Asterisk Why? Why not to 'A' B requests that call to A is closed down. The upshot of all this is that B is correctly out of the loop, and that Both A and C think they have opened communications with a new phone, both via Asterisk. Unfortunately there is no Audio. If one of the parties hangs up, the connection is correctly closed. I am curious why Asterisk would put a From: of B in the final INVITE to bridge the calls. Perhaps this is just how SIP spoofs the communication so that C does not need to know about the 2 callers? Is there some way I can track down where my audio is going? As mentioned, this problem only seems to occur if A,B,C are all SIP phones, but not if A is an ISDN call. Thanks, Steve ___ Upgrade your snom firmware to the latest and make sure break key = off in the setup. Use the transfer feature in asterisk for attended transfers. ___ 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] wiki server session limit?
It seems that the wiki pages at www.voip-info.org are not responding, and this has happened before. Responds to ping but not http requests. Is there a session limit on the web site? Is it too low? Maybe another explanantion? Anyone else notice? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
You are aware that DSL (even SDSL) is half duplex and a T1 is full duplex, right? 1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out 1.5 in. a T1 will do 1.5 in and 1.5 out sustained. This is due to a separate transmit and receive path on a t1 and a shared path on sdsl. The s in sdsl means symmetrical, not duplex, that is that the signaling rate is the same in either direction, but still half duplex. For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex nature of the traffic, unlike most internet that is download-centric. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and only costs around $100 per month. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Friday, June 10, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: Barton Fisher wrote: I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart - - -- ___ 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 Where are you located? What CLEC gives you a T-1 for $290? FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
OK, You guys have me second guessing my training and experience in this area, so; 1. If I am wrong I apologize to the group. 2. I have been trying for a few minutes to find confirmation either way. From what I know about the modulation techniques used by DSL (DMT, CAP, QAM) it is impossible for the transceiver in the device to transmit and receive at the same time (unless there is discreet channels for each path and a very good transceiver). Does anyone have any definitive technical resources confirming that any form of xDSL technology can transmit and receive at precisely the same time (not interleaved). Can anyone provide a more logical explanation of why the outbound latency on every DSL modem tested increases with inbound traffic? Even at rates well below the maximum data rate, Not the case on a T1. My explanation is that the additional latency is due to packet scheduling and queuing mechanisms required by the technology. Maybe I will learn something this evening. Damon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Are you sure? Everything I have seen says SDSL = Full Duplex. That being achieved by dropping the pair that provided voice and using it for signalling. Where ADSL utilizes unoccupied frequencies and averts conflict with analog voice frequencies, SDSL takes over the whole line. SDSL eliminates analog voice capabilities in favor of full-duplex data transmission. No splitter, no analog voice-nothing but data. As a decent alternative to T1, SDSL has gotten a fair amount of attention from Competitive Local Exchange Carriers. Excerpt from http://www.isp-select.com/SDSL.htm Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, June 13, 2005 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? You are aware that DSL (even SDSL) is half duplex and a T1 is full duplex, right? 1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out 1.5 in. a T1 will do 1.5 in and 1.5 out sustained. This is due to a separate transmit and receive path on a t1 and a shared path on sdsl. The s in sdsl means symmetrical, not duplex, that is that the signaling rate is the same in either direction, but still half duplex. For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex nature of the traffic, unlike most internet that is download-centric. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and only costs around $100 per month. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Friday, June 10, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: Barton Fisher wrote: I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart - - -- ___ 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 Where are you located? What CLEC gives you a T-1 for $290? FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Notice below that the only forms of DSL touted by anyone as replacements for full duplex 1.544mbps T1 lines is HDSL. Telcos regularly use HDSL as replacements for traditional DS1 service wher the line distances are VERY short, in most cases the HDSL circuit requires 2 pair, in some very short line distance installs it can be doen with one pair. HDSL interfaces cost 5 times what typical DSL interfaces cost. If you could get full 1.544mbps full duplex emulation off of sdsl why is there no one doing it? Sure would be a lot cheaper to provision. Have you ever gotten 22 simultaneous g.711 calls to run on a 1.5mbps dsl line? (I said 22, not 24, leaving 128kbps for signaling and protocol overhead). My experience and testing sets the real would limits to 20 to 21 on a T1 and 10 to 12 on a 1.5M SDSL circuit. I am not sure that I am that wrong in my original reply, unless there is someone here with more definitive technical references. Never one to take on an entire mailing list, I just want to be sure the next time I spout off about DSL I am certain I know what I am talking about. Live and learn, right? Damon Q: What are the various types of xDSL? A: There are several forms of xDSL, each designed around specific goals and needs of the marketplace. Some forms of xDSL are proprietary, some are simply theoretical models and some are widely used standards. They may best be categorized within the modulation methods used to encode data. Below is a brief summary of some of the known types of xDSL technologies. ADSL Asymmetric Digital Subscriber Line (ADSL) is the most popular form of xDSL technology. The key to ADSL is that the upstream and downstream bandwidth is asymmetric, or uneven. In practice, the bandwidth from the provider to the user (downstream) will be the higher speed path. This is in part due to the limitation of the telephone cabling system and the desire to accommodate the typical Internet usage pattern where the majority of data is being sent to the user (programs, graphics, sounds and video) with minimal upload capacity required (keystrokes and mouse clicks). Downstream speeds typically range from 768 Kb/s to 9 Mb/s Upstream speeds typically range from 64Kb/s to 1.5Mb/s. ADSL Lite (see G.lite) CDSL Consumer Digital Subscriber Line (CDSL) is a proprietary technology trademarked by Rockwell International. CiDSL Globespan's proprietary, splitterless Consumer-installable Digital Subscriber Line (CiDSL). EtherLoop EtherLoop is currently a proprietary technology from Nortel, short for Ethernet Local Loop. EtherLoop uses the advanced signal modulation techniques of DSL and combines them with the half-duplex burst packet nature of Ethernet. EtherLoop modems will only generate hi-frequency signals when there is something to send. The rest of the time, they will use only a low-frequency (ISDN-speed) management signal. EtherLoop can measure the ambient noise between packets. This will allow the ability to avoid interference on a packet-by-packet basis by shifting frequencies as necessary. Since EtherLoop will be half-duplex; it is capable of generating the same bandwidth rate in either the upstream or downstream direction, but not simultaneously. Nortel is initially planning for speeds ranging between 1.5Mb/s and 10Mb/s depending on line quality and distance limitations. G.lite A lower data rate version of Asymmetric Digital Subscriber Line (ADSL) was been proposed as an extension to ANSI standard T1.413 by the UAWG (Universal ADSL Working Group) led by Microsoft, Intel, and Compaq. This is known as G.992.2 in the ITU standards committee. It uses the same modulation scheme as ADSL (DMT), but eliminates the POTS splitter at the customer premises. As a result, the ADSL signal is carried over all of the house wiring which results in lower available bandwidth due to greater noise impairments. Often a misnomer, this technology is not splitterless per se. Instead of requiring a splitter at customer premises, the splitting of the signal is done at the local CO. G.shdsl G.shdsl is an ITU standard which offers a rich set of features (e.g. rate adaptive) and offers greater reach than many current standards. G.shdsl also allows for the negotiation of a number of framing protocols including ATM, T1, E1, ISDN and IP. G.shdsl is touted as being able to replace T1, E1, HDSL, SDSL HDSL2, ISDN and IDSL technologies. HDSL High Bit-rate Digital Subscriber Line (HDSL) is generally used as a substitute for T1/E1. HDSL is becoming popular as a way to provide full-duplex symmetric data communication at rates up to 1.544 Mb/s (2.048 Mb/s in Europe) over moderate distances via conventional telephone twisted-pair wires. Traditional T1 (E1 in Europe) requires repeaters every 6000 ft. to boost the signal strength. HDSL has a longer range than T1/E1 without the use of repeaters to allow transmission over distances up to 12,000 feet. It uses pulse amplitude modulation (PAM) on a 4-wire loop. HDSL2 High Bit-rate Digital
RE: [Asterisk-Users] SIP Authentication
Title: Message Race, Are you saying that the default is autocreatepeers=yes? I was under the impression that the default is no and yes must be explicitly defined. Same holds true for insecure=, default no, optional yes or very. Please tell me I am not mistaken so I do not feel compelled to review a years worth of telecom bills line by line J Damon Greetings, You have stumbled on to one of the most troublesome flag for newbies; autocreatepeer. http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+autocreatepeer in your sip.conf file add a line in the [general] section autocreatepeer=no Now people can only use your Asterisk SIP connection if you create a peer entry for them in your sip.conf file. Your sip.conf file should be located in /etc/asterisk directory. cd /etc/asterisk vi sip.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Listen to multiple ports
Are you in the USA? If so call the FCC, they do not like port 5060 blocking (or any other VoIP port blocking) See here: http://www.google.com/search?hl=enq=fcc+fine+voip Not the technical answer you are looking for but the RIGHT answer. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Prepaid Sent: Monday, June 13, 2005 10:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP Listen to multiple ports Hello all I'm trying to get my asterisk config to listen to multiple ports. This is since some clients have port 5060 blocked by their ISP. Does anyone know how to do this in sip.conf or if it is even supported? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB
Search for asterisk realtime at www.voip-info.org Answer is yes, mysql or odbc. Requires head, not stable. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili Sent: Monday, June 13, 2005 11:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB Hello, I have one question regarding *. Default configuration for asterisk is to keep configuration(s) in ordinary text based config files. My question is simple: is it possible to keep those config info (at least, to start from - sip.conf, extensions.conf and voicemail.conf) on a database, which asterisk access via ODBC. If it is possible, I'd appreciate if someone points me where I can read more about it and shows me some examples. Also I'd like to know, how asterisk behaves (in terms of stability and performance) in this environment. I.N. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB
As far as performance, * caches static config, but queries realtime configs, so scalability must be impacted, but I personally have not approached the limits of realtime yet. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili Sent: Monday, June 13, 2005 11:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB Hello, I have one question regarding *. Default configuration for asterisk is to keep configuration(s) in ordinary text based config files. My question is simple: is it possible to keep those config info (at least, to start from - sip.conf, extensions.conf and voicemail.conf) on a database, which asterisk access via ODBC. If it is possible, I'd appreciate if someone points me where I can read more about it and shows me some examples. Also I'd like to know, how asterisk behaves (in terms of stability and performance) in this environment. I.N. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB
Not as fallback, but both can be used together; You can have some static info in the test files and some in realtime, * will use the sum of both. The main benefit to RT is the reduction in the need to reload Read the wiki, these answers and much more are in there. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili Sent: Monday, June 13, 2005 11:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB When in realtime mode, does * uses static configs at all? Is it possible to operate in realtime mode and have static configs (which are build based on information taken from DB) as fallback solution? I.N. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, June 13, 2005 10:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB As far as performance, * caches static config, but queries realtime configs, so scalability must be impacted, but I personally have not approached the limits of realtime yet. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on hold for agents and queues
I have an issue where when an agent answers a call from a queue and places the caller on hold the caller hears no MOH and the agent hears congestion. When a call is placed on hold that is not from a queue MOH works fine. The hold is the SIP hold feature on the phone, not a park. the music on hold is set to default for agents, queues. Has anyone experienced this? ___ 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] DTMF in Voicemail
Is anyone aware of any fixes to DTMF in voicemail after CVS head 11/15/04. I have seen a few other posts about dtmf failing in voicemail and it seems in a least one other post the CVS date was around 11/04. We use snom phones with cvs 11/15/04 dtmfmode=rfc2833 If there are fixes an upgrade would be the way to go, but everything else works now so I do not want to move forward on a newer cvs an introduce other issues. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Overheard conversation. Comments please !
Do you have a firewall between * and the internet? Have you limited the IP address ranges that have access to * Did you determine if the other call center uses the same telco, may the telco has an issue. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Asterisk Sent: Thursday, April 14, 2005 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Overheard conversation. Comments please ! Importance: High I've just been informed of a disturbing event at our call center. We run 20+ agents taking inbound and outbound calls through the queue system, and use chan_spy to monitor ongoing conversations. My supervisor received a phone call from another call center (nothing to do with us, in fact they are 200 miles away) stating that they overheard a conversation between ourselves and one of our customers that we were speaking to at the time. He was able to give reference numbers and names, (and financial circumstances) so he obviously did hear this conversation. We are running CVS head as of 10 days ago, using TE410p on 32channel ISDN primary line. Has anyone else ever heard of something like this happening ? My boss is going apeshit talking about the DPA (data protection act) and wants answers like yesterday. Quite frankly, I have no idea on where to start to look for something like this. Julian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Overheard conversation. Comments please !
What kind of voip phone? Is it possible the user conferenced 3 calls inadvertently? Easy to do on some multi call appearance phones (snom in particular) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Asterisk Sent: Thursday, April 14, 2005 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Overheard conversation. Comments please ! Importance: High I've just been informed of a disturbing event at our call center. We run 20+ agents taking inbound and outbound calls through the queue system, and use chan_spy to monitor ongoing conversations. My supervisor received a phone call from another call center (nothing to do with us, in fact they are 200 miles away) stating that they overheard a conversation between ourselves and one of our customers that we were speaking to at the time. He was able to give reference numbers and names, (and financial circumstances) so he obviously did hear this conversation. We are running CVS head as of 10 days ago, using TE410p on 32channel ISDN primary line. Has anyone else ever heard of something like this happening ? My boss is going apeshit talking about the DPA (data protection act) and wants answers like yesterday. Quite frankly, I have no idea on where to start to look for something like this. Julian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Overheard conversation. Comments please !
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Patrick May Sent: Thursday, April 14, 2005 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Overheard conversation. Comments please ! On Thu, Apr 14, 2005 at 08:13:51PM +0100, Asterisk wrote: I've just been informed of a disturbing event at our call center. We run 20+ agents taking inbound and outbound calls through the queue system, and use chan_spy to monitor ongoing conversations. My supervisor received a phone call from another call center (nothing to do with us, in fact they are 200 miles away) stating that they overheard a conversation between ourselves and one of our customers that we were speaking to at the time. He was able to give reference numbers and names, (and financial circumstances) so he obviously did hear this conversation. We are running CVS head as of 10 days ago, using TE410p on 32channel ISDN primary line. Has anyone else ever heard of something like this happening ? My boss is going apeshit talking about the DPA (data protection act) and wants answers like yesterday. Quite frankly, I have no idea on where to start to look for something like this. Julian. ___ Not a Telco guru, but I've seen that happen, though it was on an analog line here in the states at home. They were wet, or something like that, according to PacBell. We could hear a neighbor (well, they lived probably 500' away, and around a corner) and they could hear us. It kept getting worse to the point that it was like all 4 people were in a conference call. Initiating a new call didn't help. Patrick ___ The user stated that the line is PRI ISDN, not likely to be a physical short as that would take the digital line out, not produce crosstalk, had to be a switching issues with the telco or *, or user (agent) error. ___ 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] NENA CAMA Trunks for 911 and *
Has anyone ever explored what would be required to enable * to produce NENA standard CAMA signaling for interconnection with conventional e911 services? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Getting a good deal on a PRI
Call XO www.xo.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of snacktime Sent: Thursday, April 07, 2005 5:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Getting a good deal on a PRI We have 10 incoming POTS lines to our offices, and a nortel norstar pbx. I've been looking at replacing it with * at some point in the future, and the point that looks most cost effective is when we move to PRI. Problem is, I'm not really sure how to go about getting a good deal, or what questions to ask. 90% of calls will be inbound. I called up Qwest and they quoted me $800 month. I haven't called up any CLEC's yet to see what they can do. Any suggestions? We are in Seattle, Washington. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI card and TDM400P in same box
A word of caution, we ran that same setup for a while and then bagged the TDM400P in favor of 2 Sipura SPA2000 ATAs. The TDM400P kept locking up and the SPA2000 never has. No problems getting fax from * to the SPA2000 via g.711 over a FastE LAN. I am not sure if the TDM400P has gotten any better since then (last November). The PRI card has been solid. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Brown Sent: Friday, April 08, 2005 9:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] PRI card and TDM400P in same box I have an installation next week. This asterisk box has a PRI card (for the inbound PRI) and a TDM400P with 3 FXS cards in it (for 2 fax machines and a credit card machine) What do you have to do to get * to see the TDM400P? It sees the PRI card and associated channels but I cant get the TDM400P to work no matter what mix of channel numbers I use ztcfg doesnt like it. Thanks for the help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Google Group?
What I'm still wondering about is, while you can post to that group, whether your postings are actually propagated to this list. Did anybody try that? Regards, Bruno. ___ Postings to google are not mirrored here, tried it. I think we are going to start seeing many people new to * using the google group and not getting the benefit of the infinite wisdom here. I can not imagine how you would sync them, that would only result in a circular posting nightmare. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Google Group?
Why? I'd say it's only a config issue. As long as the google group has this mailing list as it's only feed and posting to the group is equivalent to posting to the list everything should be fine. How do you propose getting posts from google to here? Email? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Google Group?
Why? I'd say it's only a config issue. As long as the google group has this mailing list as it's only feed and posting to the group is equivalent to posting to the list everything should be fine. How do you propose getting posts from google to here? Email? Well, the group receives it's content by email. It's nothing else than a subscribed user. As that, it could post (email) to this list as well. Regards, Bruno. And that is where the problems starts, if the group posts via email, and is subscribed via email, you form a loop Someone posts to google, google emails the list, the list emails google, google emails the list... Am I missing something simple here? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Google Group?
Postings to google are not mirrored here, tried it. I think we are going to start seeing many people new to * using the google group and not getting the benefit of the infinite wisdom here. They will if google keeps getting the content from this list. The problem would be that people on a google group say something that may help people here and we dont see it. I have a problem with what happened last night in relation to the google groups. Google sent out emails to a couple different lists I am on (and an unknown quantity of lists I am not on) saying that the list is not subscribed to google groups. This is blatent spam from a company I used to respect and like. Googles refusal to fix the 302 redirect problem which has caused their search results to be less accurate and now basically hijacking mailing lists on a one sided basis is further cause for concern. By feeding the content of lists into their database to get people to goto their webpage and start threads there that are not shared with the list means its a fork of the list and creates less info sharing. Its a good idea that was poorly implemented. I can not imagine how you would sync them, that would only result in a circular posting nightmare. ___ google could do it if they wanted. They obviously have a subscription to the list or they wouldnt be able to get them. Well not obviously, they could be going to gmane or other mail - NNTP services that already exist. If google did subscribe to the list to get content from the list, they could easily post to the list as well. Although authentication may be an issue if a list is 'closed' and google tries to insert the email address supplied via their webpage. Regardless google could easily filter it so it only sends to the list what is entered via a webpage and not everything it gets in its 'inbox'. That would be an ideal solution so there is effectively not two repositories of information, one on google and the list which would effectively a subset of the information on their list. I do have some other issues with google groups, right after Sept 11, 2001 google decided to delete many posts that were in their groups. These were '911 propechy posts'. If they will delete for that reason, and keep it silent that they did it as well as why, what would prevent them from say not putting this post up because its not in googles best interest? What would prevent them from putting up other posts that have valid information in them just because google does not want people to be able to see the information? But now I am just ranting and I have to fix the ram in this system so I will leave this issue alone. -- Trixter http://www.0xdecafbad.com Not to mention that the Google bounce servers are on every RBL in the world. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Google Group?
On Fri, 2005-04-08 at 12:01 -0600, Damon Estep wrote: Why? I'd say it's only a config issue. As long as the google group has this mailing list as it's only feed and posting to the group is equivalent to posting to the list everything should be fine. How do you propose getting posts from google to here? Email? Well, the group receives it's content by email. It's nothing else than a subscribed user. As that, it could post (email) to this list as well. Regards, Bruno. And that is where the problems starts, if the group posts via email, and is subscribed via email, you form a loop *ONLY* if you redirect everything google receives via email back to the list. They do not have to do that they could forward only what is posted via their webpage to the list, but choose not to do (aparently) which causes a seperate list populated in part by the existing list. It creates a one way information flow to google groups but not from it. -- Trixter http://www.0xdecafbad.com Do we know who set the group up? Is that an option? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Google Group?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Friday, April 08, 2005 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Google Group? On Fri, 2005-04-08 at 12:01 -0600, Damon Estep wrote: Why? I'd say it's only a config issue. As long as the google group has this mailing list as it's only feed and posting to the group is equivalent to posting to the list everything should be fine. How do you propose getting posts from google to here? Email? Well, the group receives it's content by email. It's nothing else than a subscribed user. As that, it could post (email) to this list as well. Regards, Bruno. And that is where the problems starts, if the group posts via email, and is subscribed via email, you form a loop *ONLY* if you redirect everything google receives via email back to the list. They do not have to do that they could forward only what is posted via their webpage to the list, but choose not to do (aparently) which causes a seperate list populated in part by the existing list. It creates a one way information flow to google groups but not from it. -- Trixter http://www.0xdecafbad.com Do we know who set the group up? Is that an option? ___ It would be q heck of a lot more readable! Too bad they didn't get it right, and why use a name like asterisk-test, you would think they would have the brains to call it asterisk-users... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] unlimited iax termination
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, April 07, 2005 9:07 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] unlimited iax termination We are planning on offering unlimited IAX terminations to the US for residential/home users for USD $19.95 per month, and business users for a higher price (not yet determined), starting May 1st, but we wanted to see what kind of interest there will be for this first. If you might be interested in this, please send an e-mail to [EMAIL PROTECTED] with contact information. ___ 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 This group does not take kindly to advertisements, you have been warned! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI Advice...
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Loretitsch Sent: Thursday, April 07, 2005 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] PRI Advice... Looking for some help any way I can. I've been closely following digium's troubleshooting steps and seem to be okay there. I am connecting, via PRI, to a Definity system. When I release the board on the Definity side I get this in Asterisk: *CLI Apr 7 10:17:23 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:23 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:23 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:23 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:23 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:23 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:23 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:48 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:48 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:48 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:48 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:48 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:48 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:48 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 I've spent about 30 hours so far troubleshooting this problem to no avail with Digium and Avaya. I followed the wiki instruction to start with and have been tweaking and recompiling since then. I tried direct wiring and also using the csu from Avaya. Started with the latest from CVS and have cleaned up and gone back to the stable cvs release. Any advice would be much appreciated! The avaya side says the d-channel is out of service. Configs: /etc/zaptel.conf Loadzone=us defaultzone=us span=1,1,0,d4,b8zs bchan=1-4 #number of channels (Yes I'm only using 4 channels for now) dchan=24 #dchannel /etc/asterisk/zapata.conf [channels] context=default signalling=pri_net switchtype=national ;echocancel=yes ;echocancelwhenbridged=yes ;echotraining=400 callerid=asreceived group=1 channel=1-4 overlapdial=yes Search www.voip-info.org (wiki) I remember seeing a config posted where * and a definity were connected. ___ 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] Local Number Ports
Anyone out there (in the US) using a CLEC to do third party local number ports? Let me be more specific; Our inbound calls come in via inbound only PRIs from a local CLEC, our outbound calls go via SIP termination to a wholesale VoIP carriers softswitch. On the inbound numbers we use the carrier of record is the CLEC that we buy the PRI from, not us. When we bring a number on to our system via local number portability the number is actually ported to the CLEC that provides us the wholesale PRI. This is know as a third party LNP. Anyone doing it now? The real questions is what are you paying per number port? We have no reference for what this should cost and therefore do not know if the proposed rate is competitive and fair. Any input appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Local Number Ports
Under what type of relationship? Are you a CLEC, ITSP, or Retail Customer? Is the local carrier you mention the incumbent or a CLEC? Thanks One of our local carriers charges 17 cents per ported DID MRC, no port/non recurring charges. I've seen in the neighborhood of $15 per 10 ported numbers as an LSR charge from other carriers NRC.. and as low as 5 cents MRC per Month. I've also seen cases with no MRC per DID per month, but an NRC per number. -m On Thu, 7 Apr 2005, Damon Estep wrote: Anyone out there (in the US) using a CLEC to do third party local number ports? Let me be more specific; Our inbound calls come in via inbound only PRIs from a local CLEC, our outbound calls go via SIP termination to a wholesale VoIP carriers softswitch. On the inbound numbers we use the carrier of record is the CLEC that we buy the PRI from, not us. When we bring a number on to our system via local number portability the number is actually ported to the CLEC that provides us the wholesale PRI. This is know as a third party LNP. Anyone doing it now? The real questions is what are you paying per number port? We have no reference for what this should cost and therefore do not know if the proposed rate is competitive and fair. Any input appreciated. ___ 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 Google Group?
http://groups-beta.google.com/group/Asterisk-test Stuff shows up fast! Anyone have insight on this, did I miss something? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Google Group?
Too bad posts made to the GG do not get mirrored here... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Calvis Sent: Thursday, April 07, 2005 10:11 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk Google Group? This has some potential especially for searching. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Thursday, April 07, 2005 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Google Group? Damon Estep wrote: http://groups-beta.google.com/group/Asterisk-test Stuff shows up fast! Anyone have insight on this, did I miss something? Looks like a mirror of the mailing list... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] D Channel Becoming CORRUPTED?
Hi, This is not entirely an asterisk question but I figure someone here may know the answer to this question. On several occassions we will lose the ability to use one of our PRI lines well for our phone system anyway (we also sometimes lose PRIs on some of our access equipment, etc). After much trouble shooting I finally decide to reset the PRI (unplug it and plug it back in). This seems to fix the issue... both on the * server, as well as on our Cisco access equipment. The explination the phone company has given is that perhaps the D channel is becoming corrupted and needed to be reset. This sounds like a cop-out to me. Any thoughts? Shouldn't I be able to expect my PRI lines to run 100% without the need for a line reset? My experience has been that a provider will not fix a problematic DS1 until you identify (or help hem identify) an error condition on the line. The problems you describe are most likely caused by a high bit error rate on the lines. With very high bit error rates a device can lose sync with the line. Make sure you do not have configuration errors on your equipment, because the line does work at times, It is safe to assume you are configured at least partially correct, but make sure you pay close attention to where you are getting your timing from. In most cases a CPE device should use the line for timing. Call your Telco and tell them you would like to have at monitor placed on all of your T1s for a couple of days with a report of the bit error rate given to you at the end of the monitoring window. Ask them up front what an acceptable bit error rate is and hold them to it. You did not specify what card you terminate the * PRI with, but the Cisco device should have counters where you can see the error counts, try show interface and show controller in the privileged exec mode on the Cisco CLI. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: CRTC mandates 911/E911 for VoIP in Canada
Title: Message From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William M. Sandiford Sent: Tuesday, April 05, 2005 7:53 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: CRTC mandates 911/E911 for VoIP in Canada For those of you out there that are Canadian or otherwise interested. The CRTC (Canadian equivalent of FCC) has released its ruling on 911/E911 for VoIP providers. In a nutshell it requires all service providers that offer a fixed native exchange service to provide E911 within 90 days and all service providers offering either fixed foreign exchange or nomadic service to offer Basic 911 within 90 days. Here are the links: CRTC Decision on 9-1-1 Emergency Services forCanada http://www.crtc.gc.ca/eng/NEWS/RELEASES/2005/r050404.htm http://www.crtc.gc.ca/archive/ENG/Decisions/2005/dt2005-21.htm Anyone does not think the US FCC will not rule the same way in a matter of time is hiding from the truth, and anyone deploying * in an ITSP manner without budgeting and planning for 911 services is in for a big financial surprise. As of now (that I am aware of) * has not method of querying a selective router for the 10 digit number of the local PSAP. Our solution at this time is; Establish a relationship with the local ANI/ALI database maintainer that allows you to update location records for TNs that have been assigned or delegated to you, even if the numbers have been delegated by an upstream LEC or CLEC. There is a fee for this service. Route all 911 calls via PRI TDM circuits to a LECs or CLECs switch that already has the ability to query the selective router and route the call to the correct 10 digit PSAP number. A fee applies here too. At some point it will be necessary to build code into * that allows it to query the ALI databases directly over IP or TDM, IP will likely be the solution. When that times comes we will be happy to join in on the developers bounty as we can afford to contribute, but do not have the skills to write the code ourselves. Intrado has already built an interface that allows these queries to take place over IP, so it would just be a matter of getting an NDA executed with them, obtaining a copy of the specification, and writing the app that queries the selective router for the correct 10 digit TN when 911 is dialed. See www.intrado.com (no affiliation) -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.2 - Release Date: 4/5/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WRT54GP2A-AT
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andre Normandin Sent: Tuesday, April 05, 2005 4:36 AM To: Asterisk-Users Subject: [Asterisk-Users] WRT54GP2A-AT Hi, I've seen these Linksys wireless routers with an ATA already built in (Both for ATT and Vonage) at staples. I was wondering if anybody has one, and has been able to configure one of them for asterisk? Is the ATA portion of the router locked or can you just go into the router's webadmin pages and configure the ATA portion similiar to the way you configure the router portion? Thanks, - Andre Andre, Before you waste you time and money, consider this; 1. The ATA built into all of the Linksys voice products are simply Sipura SPA2000's, no function difference, licensed from Sipura by Cisco/Linksys. The router model just redirects 1 ethernet port to the internal SPA 2000 and hard codes QoS to that port. That is why there is only 3 Ethernet ports on the router instead of the customary Linksys 4. 2. If you really want the Linksys, you have to be an ITSP and go through the correct authorization with Linksys to buy an unprovisioned version. 3. The vonage / ATT units are locked, and I have no knowledge of anyone successfully unlocking them without the magic code. 4. Even if they could be unlocked, you are better off going legit, buy a SPA 2000 and a WRT54G, plug the spa2000 into a port on the router, and set the QoS priority on that port. Viola, you have the same functionality, with a warranty and support! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VOIP 911 Mandatory in Canada
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Taylor Sent: Tuesday, April 05, 2005 9:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VOIP 911 Mandatory in Canada http://cnews.canoe.ca/CNEWS/TechNews/TechAtHome/2005/04/05/983311.html -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956 I see that the latency on the list is resulting in duplicate posts! Well here is a duplicate reply to go with it. Anyone does not think the US FCC will not rule the same way in a matter of time is hiding from the truth, and anyone deploying * in an ITSP manner without budgeting and planning for 911 services is in for a big financial surprise. As of now (that I am aware of) * has not method of querying a selective router for the 10 digit number of the local PSAP. Our solution at this time is; Establish a relationship with the local ANI/ALI database maintainer that allows you to update location records for TNs that have been assigned or delegated to you, even if the numbers have been delegated by an upstream LEC or CLEC. There is a fee for this service. Route all 911 calls via PRI TDM circuits to a LECs or CLECs switch that already has the ability to query the selective router and route the call to the correct 10 digit PSAP number. A fee applies here too. At some point it will be necessary to build code into * that allows it to query the ALI databases directly over IP or TDM, IP will likely be the solution. When that times comes we will be happy to join in on the developers bounty as we can afford to contribute, but do not have the skills to write the code ourselves. Intrado has already built an interface that allows these queries to take place over IP, so it would just be a matter of getting an NDA executed with them, obtaining a copy of the specification, and writing the app that queries the selective router for the correct 10 digit TN when 911 is dialed. See www.intrado.com (no affiliation) ___ 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
