[asterisk-users] SIP INFO request in asterisk
Hello everybody, Do I understand correctly that Asterisk does not support sending INFO request? Here is the goal I want to accomplish and I'd be happy to hear how can it be done with asterisk. Asterisk needs to dial out and after successful call establishment it needs to send in-dialog INFO request to the callee and wait after that for another INFO message coming from callee. So call flow looks like the following: Asterisk --INVITE---Callee Asterisk--200/OK---Callee Asterisk--ACK---Callee after asterisk sends ACK it waits for several seconds and sends INFO message Asterisk--INFO---Callee Asterisk--200/OK---Callee after receiving 200/OK callee sends INFO message and terminates the call Asterisk---INFO---Callee Asterisk---200/OK---Callee Asterisk---BYE---Callee Asterisk---200/OK---Callee Now, I want not only that Asterisk sending INFO message, I want INFO message to be constructed on a particularly way: INFO message has a body and Asterisk needs to put in that body value of Tag of To: header from 200/OK received from Callee. Any idea how to accomplish this? Thanks. --i.n. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts
Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have canreinvite=yes in their SIP.CONF 2. Both UAs have same codecs 3. There are no t, T settings in Dial command. I'd like to have a confirmation from * developers about this statement. I.N. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some info about Cisco's 79xx, and Sipura's phones
Hello folks, I've did some tests with different phones and Asterisk last two days and here are some results, which I want to share with audience. Cisco's 79xx and Sipura's phones/adapters on INVITE always reply with their preferred codec. So, for example, if Cisco's/Sipura's phone has preferred_codec g729a(18) and it receives INVITE from UA which has preferred codec ULAW(0), it will always reply with g729 and ignore what is preferred codec of calling party. Also, if two UAs have canreinvite=yes in SIP.CONF, then there is no difference in which order codecs are listed. If Cisco/Sipura's UA is called, then resulted codec after re-INVITEs will be preferred codec of CALLED party. There are other UA's which reply with the preferred codec of calling party. For example, SNOM and Grandstream behave this way. Hope this helps. I.N. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts
Hello! Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have canreinvite=yes in their SIP.CONF 2. Both UAs have same codecs 3. There are no t, T settings in Dial command. I'd like to have a confirmation from * developers about this statement. I.N. ___ --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, SIP, Re-INVITEs and different contexts
Hello Olle, It is easier to turn it around: Asterisk will issue a re-invite unless there is a reason to keep the audio going through Asterisk * NAT traversal issues * Canreinvite=no * Anything that needs asterisk to listen for DTMF in call * Codecs that needs to be transcoded Ok, let's dig into this issue. Here is my test case. Asterisk and two Cisco 7960 phones: 1. Extension 2780 with IP address 192.168.128.165 in context [internal] 2. Extension 1001 with IP address 192.168.128.172 in context [testing] 3. Asterisk is in the same subnet with IP 192.168.128.50 There is no NAT. extensions.conf [internal] exten = 1001,1,Dial,local/[EMAIL PROTECTED] exten = 2780,1,NoOp() exten = 2780,2,Dial,SIP/2780|15 [testing] exten = 2780,1,Dial,local/[EMAIL PROTECTED] exten = 1001,1,NoOp() exten = 1001,2,Dial,SIP/1001|15 sip.conf [2780] context=internal type = friend secret=2780 host = dynamic dtmfmode=rfc2833 qualify=yes disallow=all allow=ulaw allow=g729 canreinvite=yes [1001] context=testing type = friend secret = 1001 host = dynamic dtmfmode=rfc2833 qualify=2000 canreinvite=yes disallow=all allow=ulaw allow=g729 Is there any questions with extensions.conf or sip.conf? Ok, let move forward. Please see my comments to the end of this message. Call from 2780 to 1001, this is short Ethereal trace: --- Source Destination Protocol Info 192.168.128.165 192.168.128.50 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 192.168.128.50 192.168.128.165 SIP Status: 407 Proxy Authentication Required 192.168.128.165 192.168.128.50 SIP Request: ACK sip:[EMAIL PROTECTED] 192.168.128.165 192.168.128.50 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 192.168.128.50 192.168.128.165 SIP Status: 100 Trying 192.168.128.50 192.168.128.172 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 192.168.128.172 192.168.128.50 SIP Status: 100 Trying 192.168.128.172 192.168.128.50 SIP Status: 180 Ringing 192.168.128.50 192.168.128.165 SIP Status: 180 Ringing 192.168.128.172 192.168.128.50 SIP/SDP Status: 200 OK, with session description 192.168.128.50 192.168.128.172 SIP Request: ACK sip:[EMAIL PROTECTED]:5060 192.168.128.50 192.168.128.165 SIP/SDP Status: 200 OK, with session description 192.168.128.50 192.168.128.172 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 192.168.128.165 192.168.128.50 SIP Request: ACK sip:[EMAIL PROTECTED]:5060 192.168.128.50 192.168.128.165 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 192.168.128.172 192.168.128.50 SIP/SDP Status: 200 OK, with session description 192.168.128.50 192.168.128.172 SIP Request: ACK sip:[EMAIL PROTECTED]:5060 192.168.128.165 192.168.128.50 SIP/SDP Status: 200 OK, with session description 192.168.128.50 192.168.128.165 SIP Request: ACK sip:[EMAIL PROTECTED]:5060 192.168.128.172 192.168.128.50 SIP Request: BYE sip:[EMAIL PROTECTED]:5060 192.168.128.50 192.168.128.172 SIP Status: 200 OK 192.168.128.50 192.168.128.165 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 192.168.128.165 192.168.128.50 SIP/SDP Status: 200 OK, with session description 192.168.128.50 192.168.128.165 SIP Request: ACK sip:[EMAIL PROTECTED]:5060 192.168.128.50 192.168.128.165 SIP Request: BYE sip:[EMAIL PROTECTED]:5060 192.168.128.165 192.168.128.50 SIP Status: 200 OK On Asterisk console: -- Executing Dial(SIP/2780-1555, local/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Executing NoOp(Local/[EMAIL PROTECTED],2, ) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1001|15) in new stack -- Called 1001 -- SIP/1001-667b is ringing -- Local/[EMAIL PROTECTED],1 is ringing -- SIP/1001-667b answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 stopped sounds -- Local/[EMAIL PROTECTED],1 answered SIP/2780-1555 -- Attempting native bridge of SIP/2780-1555 and SIP/1001-667b *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Last Msg 192.168.128.172 100175e28b2556a 00103/0 ulawTx: ACK 192.168.128.165 2780003094c2-bf 00102/00102 ulawTx: ACK *CLI show channels verbose Channel Context Extension Prio State Application Data CallerID Duration Accountcode BridgedTo SIP/1001-667b testing 1 Up Bridged Call SIP/2780-1555 1001 SIP/2780-1555 SIP/2780-1555 testing 1001 2 Up Dial SIP/1001|15 2780 00:00:17 SIP/1001-667b --- Now, call from 1001 to 2780: Source Destination Protocol Info 192.168.128.172 192.168.128.50 SIP/SDP
[Asterisk-Users] How to dial several extensions with different timeouts
Hello, I know that using it is possible to dial several channels. Question is - is it possible and if yes, how to dial several channels with different ringing timeout? I mean the following - for example when SIP/500 is dialed, I want three phones to be dialed simultaneously - 1000, 2000 and 3000. During 10 seconds all phones are ringing, next 10 second phones 2000 and 3000 are ringing and after 20 seconds only extension 3000 is ringing. If I use in dial command, then all extensions are ringing simultaneously, but ringing timeout after comma is set for all channels, am I right? 500,1,Dial(SIP/1000SIP/2000SIP/3000,30) Thanks. 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
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
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 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] 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
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] Two or more asterisk servers, shared dialplan. Please help
Hi there, I need your help. Please le me know if it is possible to have following implementation in place: Asterisk server #1 (ast1) has server SIP clients with extensions 17XX Asterisk server #2 (ast2) has server SIP clients with extensions 16XX All I need that extensions from ast1 be able to call extensions to ast2. But asterisk servers need to be used only for call signaling setup. RTP must go directly between SIP endpoints. Is it possible to do? What is the best way to do it? 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] lookup for extensions on another SIP Proxy
I've got * registered with 50 SIP extensions. There are two another SIP proxies. I'd like to configure following: 1. Call from outside comes on *. * looks up for an extension 2. If no registered extension is on *, then request is forwarded to SIP proxy 1. 3. If client in not found on SIP Proxy 1, then * forwards request to SIP Proxy2 4. If client is not found SIP Proxy 2 congestion tone is generated. What is the best way to do it? 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] Display SIP useragents
Is there a way to display registered SIP useragents and sort them from CLI? 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
RE: [Asterisk-Users] Asterisk@home backup/restore question
As far as I understand, it is in .gz file. Based on what schedule backup is performed? Is it changeable? How to set more then one backup destination? For example, over the net? How do I selectively restore files? For example, I do want to restore only sip.conf and leave everything? Do I have to do it manually? I.N. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Manny A. Wise Sent: Sunday, May 15, 2005 1:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] backup/restore question @home do that for you everyday...;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili Sent: Sunday, May 15, 2005 2:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] [EMAIL PROTECTED] backup/restore question Hello, How do I routinely backup all necessary configuration files on [EMAIL PROTECTED] Is there any procedure/tool/script for it? And if I need to move * with existing configuration on a new hardware, what is the best way to do it? Thanks 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 ___ 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] Several questions. Please help
Hello, Question #1: I have * with g729 installed, and two phones - Cisco 7960 and Cisco 7905. If g729 is the only available codec for 7905's configuration, then call from 7960 to 7905 goes without any problem and both phones use g729. But if I call from 7905 to 7960 the following is displayed on * console: WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is not codec1 = 4, cannot native bridge. And * does transcoding from g729 to g711. Both phones have reinvite turned on. Why everything works only way and does not work other way? Question #2: What approach should be used to have an * as a MoH server? For example, I want to have 100 simultaneous SIP calls. The only destination of SIP calls are MoH. In a hypotactic scenario could be a case when each call requests deferent file, or 50 calls request the same file, 20 calls - another file and the rest - each individual files. So question is following - if I want to use Asterisk for this purpose, on what should I focus to? If all files are on the same server where * is installed, then in which format they should be stored, if a) only g711 codec is used and b) speech is 90% of each individual file? What player should I use for this purpose? Where do you see a resource bottleneck - CPU, disk system or something different? If having sound files on Asterisk server is a bad idea, where should be they stored? Question #3 How do I see ongoing transcoding session done by Asterisk from CLI? Question #4 How do I configure the following situation: Call comes in extension 555. While extension 555 is ringing extension 444 picks up. 555 continues ringing until someone picks up and in this moment call is automatically transferred from 444 to 555. 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] Asterisk@home backup/restore question
Hello, How do I routinely backup all necessary configuration files on [EMAIL PROTECTED] Is there any procedure/tool/script for it? And if I need to move * with existing configuration on a new hardware, what is the best way to do it? Thanks 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] Strange problem with G711/G729, Cisco and Grandstream
Hello everybody, Further interesting details about BT-100, * and Cisco 7960. Asterisk has G729 installed, on BT-100 there is g729 selected on all codec selections. On Cisco 7960 preferred codec is g711. Form sip.conf [1707] ;- Cisco 7960 context=default type= friend username=1707 host = dynamic dtmfmode=rfc2833 qualify=2000 disallow=all allow=g729 allow=ulaw [3710] ; - GrandStream Bt-100 context=default type=friend username=3710 user=phone host=dynamic dtmfmode=rfc2833 [EMAIL PROTECTED] qualify=2000 disallow=all allow=g729 allow=ulaw When 7960 calls BT-100 there is g729 used on both ends. sipsrv1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Last Msg 67.126.23.2513710118e46ce79a 00103/0 g729Tx: ACK 192.168.128.171 170700070ef7-36 00102/00101 g729Tx: ACK But when BT-100 calls 7960 the following is happening: -- Executing Dial(SIP/3710-8f2b, SIP/1707|15) in new stack -- Called 1707 -- SIP/1707-e96a is ringing -- SIP/1707-e96a answered SIP/3710-8f2b -- Attempting native bridge of SIP/3710-8f2b and SIP/1707-e96a May 4 16:46:58 WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is not codec1 = 4, cannot native bridge. sipsrv1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Last Msg 192.168.128.171 170702fff7f7169 00102/0 ulawTx: ACK 67.126.23.2513710b5d3f977ea1 00101/52181 g729Rx: ACK When this bug is gonna be fixed? 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
RE: [Asterisk-Users] asterisk call generator
Only signaling or with media stream also? You need commercial hardware platform. Those cost ~$20-100K. Probably you can rent those boxes. I do know, that Spirent Communications has boxes for SIP/H323/Skinny. I.N. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam Njenga Sent: Thursday, April 28, 2005 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] asterisk call generator Hi all Am looking for a way to generate like 300 simultanious calls to test *'s perfomance on a big load. * is currently working perfectly with H323, sip and IAX. Any suggestions are welcome Sam Njenga ___ 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] Is there any chance to bring Skype and AsteriskUser together?
What do you mean? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki Sent: Tuesday, May 03, 2005 3:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Is there any chance to bring Skype and AsteriskUser together? Hi, is there any chance to bring Skype and Asterisk User together? Regards, Kib ___ 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] Thanscoding and MoH questions
If there is another MoH source what is the correct way to use it with extensions? Let me explain it - asterisk has MoH on extension 555. Call comes on extension 111, so asterisk should connect incoming call to extension 555 until someone answers on extension 111. Second question: if there is a transcoding going on, how do I see detailed information about it - peers involved, extensions, IP addresses, ports, codecs from/to and so on from CLI? Thanks, 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
RE: [Asterisk-Users] Install Asterisk on CCM MCS-7835 Server
This is an ordinary HP/Compaq/IBM server. You can install * on those servers and install CCM on a ordinary computer with Intel chipset without much problems. I.N. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid Azab Sent: Thursday, April 28, 2005 1:08 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Install Asterisk on CCM MCS-7835 Server Hi All, I am replacing Cisco Call Manager with Asterisk. As you know CCM is on a MCS 7835 Server which comes with a custom version of Windows. Does any one know how to install Linux on that H/W. My guess is that someone must have tried the same thing before. I know how to install Linux however I cannot get passed the H/W limitation. Walid ___ 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] Grandstream, Asterisk and codec mismatch
Here is the situation. I've got * installed. I have Grandstream BT-100 with latest beta firmware installed and Cisco 7960G. [3710] ; - Grandstream context=default type=friend username=3710 user=phone host=dynamic dtmfmode=rfc2833 [EMAIL PROTECTED] qualify=2000 disallow=all allow=g729 allow=ulaw [1707] ;- Cisco 7960 context=default type= friend username=1707 host = dynamic mailbox = [EMAIL PROTECTED] dtmfmode=rfc2833 qualify=2000 disallow=all allow=g729 allow=ulaw On Cisco 7960 preferred codec is G711. Here is a fragment from sipdefault.cnf - # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: g711ulaw - On BT-100 in codec list #1 and #2 are G729 and then G711, G723.1 and so on. When 7960 calls BT-100 then G711 codec is used. But when BT-100 calls 7960 always G711 is used. My question is - there any way to force using G729 codec when BT-100 calls 7960 with following conditions: 1. without setting G729 as a preferred codec for 7960 2. without setting all codecs in BT-100 to G729? So, is there a way for a outgoing call from BT-100 to use local end's preference of codecs, instead of remote end? 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] Asterisk on a media stream vs. direct RTP communication between endpoints
Hello everybody, I'd like to know was there any load tasting done with *? Let's imagine 500 SIP clients on a server, 80 simultaneous calls. No transcoding, G711 or G729 codecs are used between endpoints. How asterisk performs with 80 simultaneous calls when it sits on a media stream? Is there any recommendation for hardware? Is there any graphs available showing degradation of performance or adding latency on a same hardware when number of simultaneous calls increases? Anybody? Thanks, Irakli P.S. The reason for this question is that I try in my VoIP designs to eliminate central point for RTP streams. And so far I'm convinced that a correct resign requires direct RTP communication between endpoints. ___ 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] Any other MoH source except *
If there is another MoH source what is the correct way to use it with extensions? Thanks, Irakli ___ 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, Asterisk and NAT
Hi there, There are plenty of good documents on Asterisk, SIP and NAT on the voip-info.org wiki. Please look them up. There are also information within the configs/sip.conf.sample file within Asterisk. Folks, let's face it - documentation on Asterisk sucks big time. This is the reason why the same questions are asked here and over the Net every week. If Asterisk is on a public IP, again: it's up to the phones. It's still not an Asterisk problem. Yes, but you need to pick the right phone, the right NAT/FW and have a lot of patience :-) OK, let's document is. Is there any information with different phones/FW combinations already available? I can add some info working with Cisco's IP phones, Pix firewall, cheap linksys/dlink gateways. Good NAT traversal support. and we do not give them any NAT traversal support. Why? Is this for some political reasons? Irakli ___ 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] Transcoding times
On what trascoding time depends on? I started server, run * and issued command show translations -- sipsrv1*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - 3 2 2 2 2 1 358 -61 gsm53 - 2 2 2 2 1 358 -61 ulaw53 3 - 1 2 2 1 358 -61 alaw53 3 1 - 2 2 1 358 -61 g72653 3 2 2 - 2 1 358 -61 adpcm53 3 2 2 2 - 1 358 -61 slin52 2 1 1 1 1 - 257 -60 lpc1054 4 3 3 3 3 2 -59 -62 g72955 5 4 4 4 4 3 5 - -63 speex - - - - - - - - - - - ilbc54 4 3 3 3 3 2 459 - - -- If I restart * and issue the same command -- *CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - 3 2 2 2 2 1 315 -14 gsm14 - 5 5 5 5 4 618 -17 ulaw11 3 - 1 2 2 1 315 -14 alaw11 3 1 - 2 2 1 315 -14 g72611 3 2 2 - 2 1 315 -14 adpcm11 3 2 2 2 - 1 315 -14 slin10 2 1 1 1 1 - 214 -13 lpc1012 4 3 3 3 3 2 -16 -15 g72913 5 4 4 4 4 3 5 - -16 speex - - - - - - - - - - - ilbc12 4 3 3 3 3 2 416 - - -- Why? Irakli ___ 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] Questions about ongoing calls
Two questions. If there is a VoIP-VoIP call, how do I see from a console what codecs are in use by peers? Second question: if there is transcoding going on, how do I see detailed information about it - peers involved, extensions, IP addresses, ports, codecs from/to and so on? Thanks, Irakli ___ 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] SIP, Asterisk and NAT
100k question - does asterisk correctly handle following situations: 1. Asterisk is on a public IP Two SIP clients on separate networks, each of them are behind dynamic NAT gateway. Nat gateway does not have ALG. Media stream SHOULD NOT go thought asterisk. 2. Even worst case - three clients, two of them on one site, second is on another site. For example extensions 500 and 600 are on the same site and in the same subnet and extension 1000 is on another site/network. There are PAT FW/gateways with dynamic public IP in front of clients and those are symmetric NAT/FW. The task - clients registering on Asterisk server, calling each other and RTP should not go via asterisk. So, media stream should go directly from one client to another. I want to know: 1. Is it possible? - yes/no. Implementation should involve asterisk and SIP clients and not involving third party hardware products - ALG, session border controllers or so on. 2. If it is possible, what are requirements for SIP clients. 3. What configuration changes should be done on Asterisk server and on a sip clients. And final question - if it is NOT possible with Asterisk, do you know an open source product which works in above stated scenarios and you've actually tested it. Thanks for your help. Irakli ___ 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] Manipulation based on SIP extension
Hello there, How do I configure any type of action based caller's extension and dialed number? For example if someone on extension 1777 calls extension 1777 this should be treated as accessing his voicemail box, so he won't need to call voicemail and entering mailbox number and password. 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
Re: [Asterisk-Users] Music On Hold and ATA-186 w/Silence Supression
Hello, Alejandro! AG I have a problem with ATA-186 configured for silence supression Don't! 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
Re: [Asterisk-Users] really small box
Hello, Matt! MR fine. If you have to do any sort of transcoding a soekris is not the MR way to go but for a small installation it works great. Well.. Cisco's 17xx series router is a device which you can take, plug, configure and have office PBX. But price tag is $2K. Why the same can't be done for a fraction of this price using * and not involving active cooling and graphics cards? 20-30 office users + 3-4 transcoding sessions + voicemail. What kind of horsepower do you need for this? MR I run an entire asterisk installation off of a 512 MB CF card (have MR ~250 MB to spare for voicemails and logs) Do you have install/configuration/HOWTO document? If yes, could you post it here or just send it to mail email? 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
Re: [Asterisk-Users] Router with QoS recommendations
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, April 03, 2005 3:33 AM Subject: [Asterisk-Users] Router with QoS recommendations As I have a Cisco PIX 515, with NO QoS functionality, and I'm looking for a router that does outgoing QoS to put in front of my PIX. PixOS 7.0.1 supports QoS. Yesterday it was on TAC's download page. No, I have not installed yet. 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] really small box
I don't know following has debated here or not, but is there in this world following stuff: A small, physically small box, like cable/DSL router, which comes with: 1) Ethernet port, 2) Console port, 3) CompactFlash or USB port, 4) memory module port, like SODIMM Box has built-in flash (256MB or 512MB) with or without Linux and feature to upgrade built-in RAM (128/256M) by adding memory module and storage via CompactFlash/USB. Box should have inexpensive x86 CPU in 500Mhz-1Ghz range without active cooling and should not have VGA port. It also should not have price tag more then $200. Anybody have seen stuff like this? Linksys NSLU2 and MacMini are not an option. 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
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Hello, Olle! OEJAsterisk 2.0 was moved to a Microsoft platform due to the OEJdemand for higher stability and a more secure foundation. Nice... I remember that about 10 years ago, when I was working in a daily newspaper we wrote and article on April 1st on a first page about scientific breakthrough with lunching new satellite. Satellite was going to transmit energy and electricity from space directly to homes of million customers... We've got pretty interesting calls that day, including from some low enforcement officials... 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
RE: [Asterisk-Users] Problem with Music on Hold. Please help
Any idea when this gonna be fixed? -Original Message- From: Kanuri, Seshu (Company IT) [mailto:[EMAIL PROTECTED] Sent: Thursday, March 31, 2005 7:20 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problem with Music on Hold. Please help I am having similar issue with Build 1.0.7 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, March 30, 2005 9:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with Music on Hold. Please help Hello everybody, I've run on a problem with music on hold. Asterisk does not play anything. ___ 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] Echo on internal SIP
Rule of thumb - echo is caused by remote node. Check other end. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philip Siegrist Sent: Thursday, March 31, 2005 7:37 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Echo on internal SIP Hi All, On my * server I am getting echo on internal SIP calls. I.E. Sip 2 Sip. Calls going over the T1 via the T100p are fine. I have used ulaw and gsm, gsm has less echo but it is still noticable. All phones are snom 190s. Any ideas on what i can do to cancel this. 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] Problem with Music on Hold. Please help
If you are using FC4, I think you are using Kernel 2.6, in which case usb is not needed. Anyway, is there a cure for this problem? ___ 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] Problem with Music on Hold. Please help
Hello everybody, I've run on a problem with music on hold. Asterisk does not play anything. Here is the info: latest Asterisk: Asterisk CVS-HEAD-03/25/05-23:18:57 Asterisk is installed Fedora Core 4 running on AMD 2.0Ghz CPU box with 512 RAM. I took latest zaptel source code, uncommented ztdummy and installed according to instruction from this blog - http://blog.soolid.it/?p=16 I have also compiled and installed Madplay according to same instructions. Zaptel has compiled successfully. Modprobe of zaptel/ztdummy is successful also. However lsmod output shows that USB controller is not used by ztdummy: [EMAIL PROTECTED] src]# lsmod Module Size Used by ztdummy 3924 0 zaptel204676 7 ztdummy ohci_hcd 23765 0 uhci_hcd 31449 0 ehci_hcd 35273 0 Asterisk starts without a problem, the only messages I've receive are following: WARNING[26256]: chan_oss.c:486 soundcard_init: Unable to open /dev/dsp: Device or resource busy == No sound card detected -- console channel will be unavailable ERROR[25489]: cdr_custom.c:135 load_module: Unable to register custom CDR handling Everything else works, but as I said there is no music on hold. sip.conf: In global parameters: musicclass=default ; Sets the default music on hold class for all an extension: [2707] context=default type = friend username = 2707 host = dynamic mailbox = 2707 dtmfmode=rfc2833 nat=no disallow=all allow=ulaw allow=g729 musicclass=default In extensions.conf file: exten = 2707,1,Dial(SIP/2707,35,trHm) ;exten = 2707,2,MusicOnHold() ;exten = 2707,3,MP3Player(/var/lib/asterisk/mohmp3/fpm-sunshine.mp3) exten = 2707,3,voicemail(u2707) exten = 2707,4,Hangup exten = 2707,102,Voicemail(b2707) exten = 2707,103,Hangup musiconhold.conf file: [classes] ;default = quietmp3:/var/lib/asterisk/mohmp3 loud = mp3:/var/lib/asterisk/mohmp3 default = custom:/var/lib/asterisk/mohmp3/,/usr/bin/madplay --mono -R 8000 --output=raw:- But it does not work. when I call extension 2707 on console is following output: Reloading SIP Urgent handler Use EXIT or QUIT to exit the asterisk console -- Executing Dial(SIP/1730-b6a4, SIP/2707|35|trHm) in new stack Urgent handler Urgent handler -- Called 2707 Urgent handler -- Started music on hold, class 'default', on SIP/1730-b6a4 Urgent handler -- SIP/2707-8c7d is ringing Urgent handler -- Stopped music on hold on SIP/1730-b6a4 Any idea what is going wrong? 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
RE: [Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here
G'day mate, I've got 15 7960/7940 in my office with firmware 7.4 and have no problems. I can make calls from the 7960. When I get a call placed to the 7960 the call is setup but there is no audio in either direction. Is call from 7960 to 7960? I have tried firmware versions 6 7 on the Cisco phones, same result. Means - something wrong is with your config... [9001] type=friend ; either friend (peer+user), peer context=extensions secret=9001 fromuser=Cisco ; overrides the callerid, e.g. required by FWD callerid=9001 host=dynamic; we have a static but private IP address nat=never ; there is not NAT between phone and dtmfmode=rfc2833 ^^^ see below, in phone section. canreinvite=no ; allow RTP voice traffic to bypass Asterisk Hmm... something tells me that RTP stream goes to asterisk, instead of phones' ip addresses. Could you check, what parameter do you have in global and second 7960's section? progressinband=yes Why do you need this? disallow=all allow=ulaw Seconf 7960 has the same config in SIP.CONF? dhcp_server : 192.168.10.254 my_ip_addr : 192.168.10.17 subnet_mask : 255.255.255.0 defaultgw : 192.168.10.254 tftp_addr : 192.168.11.2 Does phone receive sipdefault.cnf and SIP.cnf file from TFTP? dtmf_outofband : avt dtmf_avt_payload : 101 dtmf_db_level : 3 dtmf_inband : 1 You've set in Asterk's config DTMF as out_of_band, while in phone's config you'set as in_band. Corret it first. proxy1_address : 192.168.10.106 do sip debug ip ip_address_of_7960 ad take a look. Good luck! ___ 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] No prompt after installing
ntpdate ntp1.cs.wisc.edu 30 Mar 23:15:20 ntpdate[3840]: no server suitable for synchronization found time-a.nist.gov ___ 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
