Re: [asterisk-users] Default extension
On 26 Mar 2014, at 19:14, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hello, When I get a SIP INVITE as follows: INVITE sip:s@10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18 To: sip:02XX@IP:5060 Contact: sip:1053212@IP:5060 Call-ID: 344d42bd16975a54141d11f635bdf...@sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 252 Asterisk considers that the extension is 's'. (The Register) How to make the extension number that is shown in the 'To' ?? You never route calls on the To: header in SIP. You route on the request URI. Unless this is something where you used the REGISTER statement in sip.conf and forgot to add an extension or you register once for multiple DIDs. I would suggest changing your register statement to include an extension. In that extension you read the To: header with the SIP_HEADER() dialplan function and issue a goto so you end up with the extension in the To header. The IETF has with help of the SIP forum written a standard extension to SIP to handle this use-case, something called GIN. It's now part of the SIPConnect specification. using the gin extension, you would get the called phone number in the r-uri. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip.conf and templates
6 jun 2013 kl. 17:41 skrev Daniel Pocock dan...@pocock.com.au: On 06/06/13 15:51, Daniel Pocock wrote: Is the template capability in sip.conf compatible with realtime sip.conf entries such as users in a database? I notice that contrib/realtime/mysql/sippeers.sql and the wiki page don't mention a template column: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure while some third-party examples do suggest that a column named template is permitted: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip I have actually tried adding that column template into sippeers and setting the value as the name of a template from my sip.conf - on Asterisk 11.4, it seems to ignore the column. If there is a way to do this, it would be useful to have it in the wiki. The templates are part of the configuration file (text files) parser and not supported in databases. /O - SIP Masterclass in Malaga Spain - July 2013! Register now - read more at http://edvina.net/blog/2013/01/sipmaster-malaga-2013/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPv6 and IPv4 binding address on a server with 2 network cards
13 mar 2013 kl. 20:30 skrev Miguel Baptista miguel.bapti...@uninett.no: Thank you Olle. Well, in my test scenario I will leave the bindaddr=:: value. Any idea if/when asterisk will support two specific bindaddr (one for IPv4 and another for IPv6) ? I do hope that the new stack that will be in early beta in Asterisk 12 is going to handle that. To fix it for current asteirsk would be messy, so I just have the habit of running Kamailio in front to handle both these issues, as well as to add DoS protection and proper TCP/TLS support. Like many other things, these kind of issues will be fixed when there's someone that needs to fix it and provides funding for a developer to do it or have developer resources, fix it and contribute the code back to the project. /O - Miguel Baptista On 3/13/2013 10:06 AM, Olle E. Johansson wrote: ; With the current situation, you can do one of four things: ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1 ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1 ; c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0 ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=:: From the configuration file - you can do ONE of four things, but only one. You can not have multiple bindaddr= settings. /O 11 mar 2013 kl. 11:03 skrev Miguel Baptista miguel.bapti...@uninett.no: Hi Asghar, Thanks for you reply. Which Asterisk version are you using? I am using Asterisk 11.1.0 when I use the bindaddr parameters with specific IP addresses, Asterisk will listen only on the last entry. For example, when I have bindaddr=ipv4A:port bindaddr=[ipv6A]:port it will listen only on the IPv6A address and when I have the other way around: bindaddr=[ipv6A]:port bindaddr=ipv4A:port Asterisk will only listen on the IPv4A address. The only way I found to force asterisk to listen on both IPv4A and IPv6 A was to use bindaddr=[::] but it makes asterisk to listen also on the other IP addresses. Maybe this is fix on a newer Asterisk version. - Miguel Baptista On 3/10/2013 8:04 PM, Asghar Mohammad wrote: hi, i am using similer setup just put bindaddr=ipv4A:port and bindaddr=[ipv6A]:port ans it should work. On Sun, Mar 10, 2013 at 3:04 PM, Miguel Baptista miguel.bapti...@uninett.no wrote: Hello, I am doing some tests with asterisk on a dual-stack environment. I have some doubts regarding asterisk binding addresses on a server with 2 network cards. According to asterisk documentation: ; With the current situation, you can do one of four things: ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1 ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1 ; c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0 ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=:: ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for ; udpbindaddr, tcpbindaddr, and tlsbindaddr.) ; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat. ; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.) ; ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061 ; for TLS). ; IPv4 example: bindaddr=0.0.0.0:5062 ; IPv6 example: bindaddr=[::]:5062 ; ; The address family of the bound UDP address is used to determine how Asterisk performs ; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only ; records are considered. In case d), both A and records are considered. Note, ; however, that Asterisk ignores all records except the first one. In case d), when both A ; and records are available, either an A or record will be first, and which one ; depends on the operating system. On systems using glibc, records are given ; priority. Lets say that I have two network cards: A and B. Both interfaces have IPv4 and IPv6 addresses: IPv4 A, IPv6 A, IPv4 B and IPv6 B. How can I make asterisk to run only on B network addresses (IPv6 and IPv4)? The bindaddr=[::] config parameter tells asterisk to run on all available addresses, including the addresses on the A network. But that's not exactly what I want. - Miguel Baptista -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live
Re: [asterisk-users] IPv6 and IPv4 binding address on a server with 2 network cards
; With the current situation, you can do one of four things: ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1 ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1 ; c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0 ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=:: From the configuration file - you can do ONE of four things, but only one. You can not have multiple bindaddr= settings. /O 11 mar 2013 kl. 11:03 skrev Miguel Baptista miguel.bapti...@uninett.no: Hi Asghar, Thanks for you reply. Which Asterisk version are you using? I am using Asterisk 11.1.0 when I use the bindaddr parameters with specific IP addresses, Asterisk will listen only on the last entry. For example, when I have bindaddr=ipv4A:port bindaddr=[ipv6A]:port it will listen only on the IPv6A address and when I have the other way around: bindaddr=[ipv6A]:port bindaddr=ipv4A:port Asterisk will only listen on the IPv4A address. The only way I found to force asterisk to listen on both IPv4A and IPv6 A was to use bindaddr=[::] but it makes asterisk to listen also on the other IP addresses. Maybe this is fix on a newer Asterisk version. - Miguel Baptista On 3/10/2013 8:04 PM, Asghar Mohammad wrote: hi, i am using similer setup just put bindaddr=ipv4A:port and bindaddr=[ipv6A]:port ans it should work. On Sun, Mar 10, 2013 at 3:04 PM, Miguel Baptista miguel.bapti...@uninett.no wrote: Hello, I am doing some tests with asterisk on a dual-stack environment. I have some doubts regarding asterisk binding addresses on a server with 2 network cards. According to asterisk documentation: ; With the current situation, you can do one of four things: ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1 ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1 ; c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0 ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=:: ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for ; udpbindaddr, tcpbindaddr, and tlsbindaddr.) ; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat. ; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.) ; ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061 ; for TLS). ; IPv4 example: bindaddr=0.0.0.0:5062 ; IPv6 example: bindaddr=[::]:5062 ; ; The address family of the bound UDP address is used to determine how Asterisk performs ; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only ; records are considered. In case d), both A and records are considered. Note, ; however, that Asterisk ignores all records except the first one. In case d), when both A ; and records are available, either an A or record will be first, and which one ; depends on the operating system. On systems using glibc, records are given ; priority. Lets say that I have two network cards: A and B. Both interfaces have IPv4 and IPv6 addresses: IPv4 A, IPv6 A, IPv4 B and IPv6 B. How can I make asterisk to run only on B network addresses (IPv6 and IPv4)? The bindaddr=[::] config parameter tells asterisk to run on all available addresses, including the addresses on the A network. But that's not exactly what I want. - Miguel Baptista -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
Re: [asterisk-users] How does Asterisk handle ACK's?
12 mar 2013 kl. 16:54 skrev Grant Bagdasarian g...@cm.nl: Hello, I’m noticing strange behavior in one of our Asterisk nodes where the ACK is always sent to the proxy, but RR is not enabled for calls. The proxy drops the ACK. I’m using the AMI interface to originate a call: Action: login Username: myusername Secret: mypassword Events: on Action: Originate Channel: SIP/SOMENUMBER@proxy1 CallerID: SOMENUMBER Application: Playback Data: hello-world Using Asterisk 10.5.0. Shouldn’t Asterisk send the ACK directly to the endpoint in the Contact header? Yes it should. File a bug report with details. /O-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Register Free Opensips/Asterisk Integration
10 mar 2013 kl. 03:04 skrev Nick Khamis sym...@gmail.com: Hello Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an OpenSIPS/Asterisk integration. Most approaches have OpenSIPS relay the UA's REGISTER request to Asterisk which has host=dynamic set for the Friend/Peer and everything works as expected. There are a lot of models for this. Check my presentation from Astricon 2010 to get some ideas. http://www.slideshare.net/oej/astricon-2010-scaling-asterisk-installations /O Where I run into problems is in Inbound calls. When I try to call the extension from a DID I am receiving Unable to create channel of type 'SIP' (cause 20 - Unknown). And rightfully so! Reason being: SIP Show Peers Yields: Name/username HostDynForcerport ACL Port Status Realtime 1001/1001 192.168.2.5N 5060 UNREACHABLE Cached RT TTrunk/sip.exp.com 192.168.2.5N 5060 UNKNOWN Cached RT As for who will keep track of the UA location, the OpenSIPS `location` table has the correct info: select username,domain,contact,socket from location; +--+++--+ | username | domain | contact| socket | +--+++--+ | 1001 | sip.exp.com | sip:1001@192.168.2.11:5060 | udp:192.168.2.5:5060 | +--+++--+ OpenSIPS: sip.exp.com OpenSIPS: 192.168.2.5 Asterisk: 192.168.2.10 UA: 192.168.2.11 I have set `host=sip.exp.com' for the UA but the UA is still `UNREACHABLE` by asterisk As for the rest of the media related stuff, everything works perfectly. Outbound works fine. As you know, this only poses a problem with inbound calls to the UAs. Your Help is Greatly Appreciated, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF and call-limit in 1.8
6 dec 2012 kl. 16:54 skrev Danny Nicholas da...@debsinc.com: Not sure about this since I use the 10/11 branches and not 1.8, but I think you need to use the deprecated call-limit for BLF and the new busylimit for the other features you need. http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf Call-limit is the limit on the number of calls you can take and also sets a device to BUSY. Since you want to be able to transfer calls, you need at least two. But this did not set the phone to busy on one call. That's why we added busy-limit that can be set to the level you want device states to signal busy, but still give the ability to the phone to set up more calls. counteronpeer is the same as limitonpeer, just a new name. /O From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B. Christensen Sent: Thursday, December 06, 2012 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] BLF and call-limit in 1.8 Hello We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF lamps on our Polycom phones stop working. After a lot of googling and a lot of testing, I have been unable to find a solution. I did try to change the call-limit value from 4 to 1, and this actually made BLF work (noone suggested this, and what documantation I can find states that this option is deprecated). This change has other implications, however. Call waiting stops working, queues don't offer calls if the user is in a private call etc. We have customers that require both BLF and call waiting at the same time. We are running Asterisk 1.8.11-cert7 I've made the following additions to sip.conf [general]: callcounter=yes counteronpeer=yes (undocumented? Supposed to replace limitonpeers?) (old relevant values, unchanged) allowsubscribe=yes subscribecontext=blf notifyringing=yes notifyhold=yes limitonpeers=yes I also tried may other suggestions I've found like placing the hints in the same context as the extensions and removing subscribecontext. Is there something I'm missing? Is something not working correctly? Thanks in advance, Pan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and OpenLDAP
31 okt 2012 kl. 15:07 skrev Giuseppe Longo giuseppe...@gmail.com: I don't want update Asterisk configuration, i want to query LDAP only for name and secret field. Currently Asterisk can't do that. If you add Kamailio as a proxy in front of Asterisk, you can easily authenticate with LDAP this way. There was some work by Philippe Sultan in this area done years ago, but was never completed. In SIP, the MD5 Digest authentication is based on the cleartext password being available to calculate the hash. Therefore we can't use the LDAP authentication for binding as an authentication mechanism in SIP. As long as we can have a binding (authentication for the server itself) and query and in the result get a cleartext authentication username and secret, kamailio should be able to do the job. The Asterisk realtime driver assumes that you use a [peer] or [user] object like the ones we use in a database - or that you query from the dialplan with the realtime function. However, as stated earlier, this doesn't work in the SIP authentication that is based on the data in peers and users. Regards, /Olle -- * Olle E. Johansson - o...@edvina.net * Kamailio SIP Masterclass Miami FL December 2012 * http://edvina.net/training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP - Authenticated vs Unauthenticated Calls
1 nov 2012 kl. 15:13 skrev Joshua Colp jc...@digium.com: Tim Nelson wrote: Thanks Joshua- In this case, we're using SIP registration to peer the remote systems to the 'central system'. In option #1 above, the 'user' portion is always the CID we set for the outbound call, but the actual SIP user is something different like 'site12' for example. So, it would appear #1 is not a match... Registration just tells the remote system what your IP address/port is for sending calls. You don't *have* to send CID like you are. You can override using fromuser to ensure that the specific peer is *always* matched and authenticated. CID can be conveyed in an alternate header, like Remote-Party-ID. The options involved for RPID are sendrpid=yes on the caller box and trustrpid=yes on the receiving box. That leaves us with option #2. We're using 'qualify=yes' on both sides of the SIP peering. If a peer becomes unreachable (fast UDP state table timeout on a remote firewall for example) as seen by the central system, and an outbound call is made from the remote system, that would mean the call is coming from an unknown IP:port. Would this then make sense Asterisk would simply throw it into the from-sip-external context as an unknown/unauthenticated call? And of course, when the peer *is* registered, and a call is made, Asterisk on the central system allows the call as authenticated due to the source IP/port being known via the registration status? It's possible, without logs and such it's only a guess. Agree, all comments are pure speculations at this point. Try removing the user object to simplify. If you have type=friend, change to type=peer and you will *only* get IP/port-based matching and can configure your system in a controlled way. There are just a few situations where you actually benefit from having type=friend and match object names with Caller ID numbers. /O -- * Olle E. Johansson - o...@edvina.net * Kamailio SIP Masterclass Miami FL December 2012 * http://edvina.net/training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireshark AMI Dissector
23 okt 2012 kl. 22:31 skrev Kristian Kielhofner k...@kriskinc.com: Hello everyone, Does anyone know of a Wireshark AMI (Asterisk Manager Interface) dissector? Decode as telnet and display filter telnet.data kind of work but TCP reassembly can't happen without a better understanding of the protocol... No, but that's a very cool idea. Would be great to have. Cheers, /O SIP Masterclass - Miami, FL, USA Dec 2012 - register now! http://edvina.net/training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why all the 401 Unauthorized
23 okt 2012 kl. 14:28 skrev Steven Howes steve-li...@geekinter.net: Hi, SIP registrations typically try to register, are them prompted for a password (via a 401 message) it then sends a new request with authentication . This is normal. Yes it is. To be more exact and clarify: We never prompt for a password or ask anyone to send a password in clear text. SIP authentication is based on HTTP MD5 digest authentication, where we have to send an authentication challenge to get an authentication response. The challenge is sent in the first 401 and the response in the next SIP request - REGISTER, INVITE or something else - from the client. The information sent over the net is just a digest of a set of information, where the shared secret is one piece. Cheers, /O Steve On 23 Oct 2012, at 13:26, Jerry Geis wrote: I have a connection between two asterisk boxes, both running 1.4.43 The connection is alive and good and working. however, I see a bunch of 401 Unauthorized messages using wireshark - then it eventually registers again just fine. Why would it not successfully register the first time - every time? Jerry --- * Olle E. Johansson - o...@edvina.net * Kamailio SIP Masterclass Miami FL December 2012 * http://edvina.net/training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] change channel variable to a user chosen value during a call
31 aug 2012 kl. 09:18 skrev Frederic Van Espen frederic...@gmail.com: On Fri, 2012-08-31 at 00:11 +, Andrew White wrote: Is realtime an option for you to install? Andrew, Realtime is not an option actually. We have a whole system built up that generates configuration files. The primary goal is to let the user directly change the channel variable with his phone, while in conversation, or with a short interruption of the call. If that isn't possible, an AMI call will be just fine. I'd just like to make sure it is not possible on the phone itself first. There is a hidden feature for SNOM phones in the SIP channel. They have a way to send a client code during the call (made for lawyers) and that will end up in the CDR. /O smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?
31 aug 2012 kl. 13:13 skrev Steve Davies davies...@gmail.com: On 31 August 2012 07:49, Olle E. Johansson o...@edvina.net wrote: 24 aug 2012 kl. 16:18 skrev Steve Davies davies...@gmail.com: Hi SIP Gurus, I've tried to find the relevant RFCs, but am struggling. I can find the odd opinion online, but was wondering if anyone could give a definitive answer. If a SIP call is initiated (INVITE) and receives either a 180 with SDP, or a 183 with SDP, then the remote party will start to send audio for the inband-ringing. Asterisk then passes this audio, and it is correctly heard by the caller. At present, Asterisk will also start to pass back any handset audio in return, in theory allowing a conversation to occur on an unanswered channel if an endpoint were designed to allow this (free phonecalls here we come!). My question: Should: 1) Asterisk block outbound audio between the 183 Progress and the 200 OK packets? 2) Replace any outbound audio with silence? 3) Replace outbound audio with a special NULL RTP of some sort? Does that exist? 4) Allow any audio to be sent regardless? I have implemented 1) at present on our test rig, but the lack of outbound RTP causes issues with firewall state not being set-up to allow the inbound audio. I am not sure how hard/easy it would be to do 2) as you'd need to create silence of the correct duration to replace each audio frame. Thoughts please? First, because of Asterisk's RTP implementation we have to send some RTP packets at this point. You could mute the calling channel before calling and unmute the channel at answer if needed, but normally sending audio won't hurt. A normal user should not be able to send early media on a pstn-like installation where you bill the calls, so there should be little risc of two-way conversations before an answer. In some cases you have to let the caller send DTMF (the famous fed ex example) in early media, so we can't block any media by default in Asterisk. Using the r option in dial causes a lot of issues, since you can still get busy or congestion when you have early media, so that is not a good solution. /Olle Excellent information as always Olle. Many thanks. My intention is to make the early-audio prevention in SIP a little more harsh, such that if SIP receives audio before a 183 or 200 is received, it is dropped. This fixes the case where useless early-audio is received from a non-SIP (eg ISDN) technology, and can cause an onward node to auto-enable early audio mode, causing silent ringing and other broken behaviours. This is one of my pet issues. THe problem today is that many gateway vendors ALWAYS send 183 with sdp, regardless if it's a ring tone or a service provider message. If you kill the 183, service provider messages will disappear. My recommendation (which I've mentioned in tons of mails and blog entries) is to send 180 ringing with SDP for ring tones and 183 for other messages. That way I could kill the 180 SDP in a Kamailio proxy before it hits the Asterisk server. In reality today, by killing 183 you will also block important information for the caller, like this subscriber has a new number. /O * The New Edvina SIP Masterclass - Stockholm and Miami this fall! http://edvina.net/training/new-sip-masterclass/ smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API
31 aug 2012 kl. 16:58 skrev Shitian Long longst...@gmail.com: Do you think it is a good way to use Manager API command action to implement this feature? No. The command action should be avoided since the output from the CLI commands is not made for parsing by applications and may change too. Sometimes we cut of informaiton to fit into a terminal window. If you use manager actions instead, you will always get the full data in a format you can parse. If you have to use the command action you have found a place where a manager action is missing and we developers would like to know that and fix it :-) For realtime, there's a dialplan function REALTIME() that you can use with the manager actions that change or read channel variables. That's the best way, since we lack manager realtime commands. One reason for not going directly to the database API is that when building 3rd party apps, we don't know what database you are using and can benefit from the ARA interface to databases, exactly like Asterisk. It's not as effecient as going directly when you can, but sometimes you just don't know what's behind ARA and thanks to ARA you don't have to. :-) /Olle * The new Edvina SIP Masterclass - Stockholm, Sweden Oct and Miami, FL, Dec 2012 http://edvina.net/training/new-sip-masterclass/ smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requiring agent to confirm queue calls only when forwarded to external device
17 aug 2012 kl. 03:15 skrev Phillip Frost: On Aug 16, 2012, at 6:25 PM, Tiago Geada wrote: forward to a Local extension that has dialplan requiring the acknowledgement? On 16 August 2012 21:12, Phil Frost p...@macprofessionals.com wrote: I'd like to allow my users to forward their calls using the forwarding feature on their SIP handsets and continue to receive Queue() calls. Currently I set the 'i' option in Queue() so that if a user forwards to their cell phone, or any other extension that has voicemail, the voicemail doesn't eat all the calls to the queue. I'd think that would require teaching all the users to forward to a different extension if they thought they could be receiving queue calls. My users probably aren't that good at following directions ;) Ultimately, I'm sure I could solve this problem by taking management of forwarding off the phone and into Asterisk, since then I'd absolutely have some flag indicating if forwarding is active or not. However, I was just hoping there was an easier way. I'm really happy with the forwarding interface on our current handsets, and I'd rather not go through the effort of changing their configuration, or changing the user experience if I can avoid it. If a call is forwarded and hit the dialplan again, it's forwarded to the context set in the channel variable FORWARD_CONTEXT. So you could set this variable before you hit queue(), then do things differently in the context specified by this variable, since you know that the call is forwarded. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC List
8 aug 2012 kl. 14:07 skrev Kevin P. Fleming: On 08/08/2012 06:30 AM, Kannan wrote: Where can I get a complete set of RFCs and other specifications supported by Asterisk? To my knowledge there is no such list. In addition, Asterisk (like many other pieces of software) does not claim 100% compliance with every RFC that is relevant, so usually it's better to ask about the specific features you are interested in. THis is a document I haven't updated since 1.6.x but still covers a large part of the SIP implementation: http://svnview.digium.com/svn/asterisk/team/oej/sip-compliance/asterisk-sip.txt?view=markup /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf and binaddr issue
10 jul 2012 kl. 20:50 skrev Kevin P. Fleming: On 07/10/2012 03:24 AM, Olle E. Johansson wrote: The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. In fact, it's a well known bug that if you have multiple interfaces with different IP networks, Asterisk will send from the wrong IP on some of the interfaces. Are you sure about that? The only problem area that I'm aware of is when there are multiple *overlapping* interfaces (on the same subnet, or providing the same route(s)). In that case, Asterisk can receive messages on one IP address out of the overlapping set, but reply using a different one from the set, because it doesn't specify the source IP address and instead lets the UDP/IP stack select one. If the interfaces don't overlap in any way, I don't see how it would be possible for Asterisk to send messages with the wrong source IP address, since it does not specify the source IP address at all. If this is occurring, it must involve the operating system's IP stack in some fashion. Yes, I still use quite a lot of IPtables tricks to overcome this issue. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forcing SIP trunk matching order?
11 jul 2012 kl. 00:26 skrev James Lamanna: On Mon, Jul 2, 2012 at 12:13 AM, Olle E. Johansson o...@edvina.net wrote: No. This is probably because you are using phone numbers as names of devices with type=friend in sip.conf. That's generally a bad idea. The SIP channel matches an incoming call this way: 1. Take the From: user name and match with the list of type=user and type=friend 2. Take the sender's IP and port and match with the list of peers 3. Send the call to the context defined in the [general] section of sip conf In Asterisk 1.4 and hopefully 1.8 the last peer in sip.conf will match first. In 1.8 the internal strcutures was changed, but I hope that this functionality was retained. We had a dicussion about it, but I personally haven't tested the result. One needs to know the matching order, so if 1.8 doesn't behave that way, we need to fix it. The recommended way is to NOT use anything that likely will end up as a caller ID as names of devices in sip.conf. The name is whatever you have within square brackets above definitions of type=friend or type=user. The username= option is another option, not the name of the device. The quick way to solve your problems is to stop using type=friend and start using type=peer instead. Hi Ollie, You are correct, I do have callerID-type names as accounts in sip.conf. The hosts are set to dynamic. Is this a problem with type=peer? As stated above, peers only match on IP+port for incoming calls. Would the deny/allow suggestion posted earlier also work with type=friend? Deny/allow is a different thing and doesn't really affect matching. It is applied AFTER matching, not during or before. Cheers, /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timer1 RFC and SIP.CONF
6 jul 2012 kl. 09:29 skrev Elliot Murdock: Hello, Thank you for the clarification. Just a few questions: 1. What is the Timer1 used for? Timer1 is the base for many other SIP timers and it's an estimate of the roundtrip time for a packet between two SIP devices or servers. TimerB is based on T1, like the retransmit timers. 2. Since timerb is for all responses, final and provisional, the comment in sip.conf perhaps should point that out instead of implying only for provisional responses: If a provisional response is not received in this amount of time, the call will autocongest Yes, that should propably change. /O Thanks, Elliot On Thu, Jul 5, 2012 at 12:31 PM, Olle E. Johansson o...@edvina.net wrote: 4 jul 2012 kl. 13:32 skrev Elliot Murdock: Hello, I am trying to get clarity with the sip.conf timer configuration. The current configuration states: ;--- SIP timers ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;timert1=500; Default T1 timer ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 However, according to RFC 3261: (EXCERPT 17.1.1.1) T1 is an estimate of the round-trip time (RTT), and it defaults to 500 ms. Nearly all of the transaction timers described here scale with T1, and changing T1 adjusts their values. The request is not retransmitted over reliable transports. After receiving a 1xx response, any retransmissions cease altogether, and the client waits for further responses. The server transaction can send additional 1xx responses, which are not transmitted reliably by the server transaction. Eventually, the server transaction decides to send a final response. (EXCERPT 13.2.2.4 2xx Responses) The UAC core considers the INVITE transaction completed 64*T1 seconds after the reception of the first 2xx response. According to the RFC, the 64*t1 timeout is not for provisional responses, but for final responses. This seems to be in contradiction to what is stated in the sip.conf file. Unless you have PRACK support, which Asterisk not yet has, there's no timeout in the SIP protocol for provisional responses more than the need to update your provisional response at least every minute not to hit a 3 minute timeout in the SIP transaction state in a proxy. Now, the timerb is used to get ANY response from a server out there, including provisional responses. If we don't get ANY response within TIMERB, the SIP transaction dies and in a UA with a transaction layer we generate a local 408 timeout. In Asterisk, we congest the call. So the 64*T1 is for any response, including final response. It's there to decide whether or not you have intelligent SIP life forms handling your SIP request in the network universe. I hope this clears up your confusion. Regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf and binaddr issue
6 jul 2012 kl. 23:18 skrev Felix Salfelder: Hi there. i am seriously stuck with an asterisk and sip problem. the following sip.conf works with respect to some_peer: [general] bindaddr = x.y.z.w nat = no [some_peer] type=peer host=somehost secret=somesecret some other unrelated options here x.y.z.w is the ip address of the interface pointing to the network containing somehost. more precisely its the address of tun0 and route -n prints Destination Gateway Genmask Flags Metric RefUse Iface [..] x.y.z.0 0.0.0.0 255.255.255.0 U 0 0 0 tun0 [..] here 'it works' implies that i have to change and reload sip.conf after ifup tun0, or anything that forces tun0 to go down, like my dsl provider. also, the bindaddr line is suboptimal for the other peers... the same thing -- without the bindaddr part -- doesnt work. more precisely it almost works. its just incoming sound that doesnt. this must have something to do with how asterisk picks up interface addresses and communicates them to the peer in question. inspecting the packages sent to somehost, gave me the impression that asterisk uses the ip adress of ppp0 (a dsl modem) instead. how am i supposed to tell asterisk to use tun0 as the interface for [some_peer] so i can remove the bindaddr line? i've found many nat-related options in the manual, but there is no nat involved here. also, i couldnt find anything similar to iface=tun0, although the sip dialogue apparently relies on ip adresses and routing. this is about asterisk on sid, version 1:1.8.13.0~dfsg-1, but of course i'm going to switch to whatever you might suggest. The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. In fact, it's a well known bug that if you have multiple interfaces with different IP networks, Asterisk will send from the wrong IP on some of the interfaces. Sorry, /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rookie / sip and extensions
7 jul 2012 kl. 21:07 skrev Mikhail Lischuk: Thomas Perron писал 07.07.2012 21:48: exten = s,n,Dial(SIP/16175551212) sip.conf [general] ;register = 125010155:funnyti...@sip3.voipvoip.com/125010155 register = 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11 ; [incoming] username=125010155 I dont know what you are trying to do, but: 1) Peer doesn't have to be the same name as context. Change [incoming] in sip.conf to something like [voipvip] - it will be easier later when you have more peers. 2) What is 16175551212 ? You don't have such peer in sip.conf. If it's a number, Dial should be SIP/peer/number, for example SIP/voipvip/617 or whatever you want to dial 3) If you've posted your real password here - I strongly suggest you change it right now Please note that the account name is the name within square brackets. The username= option (now renamed to defaultuser= ) is a very different thing, and NOT the username of the account. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] seems like call is picked and returned to me
9 jul 2012 kl. 15:24 skrev Sergio Serrano: Hi all I hope that someone of you can solve this. Right now I'm stuck! I'm using asterisk with some SIP extensions. Basically I want to establish a call between desktop voip phone (ext 181) and embedded sip system (ext 182) All I can see in CLI is: == Using SIP RTP CoS mark 5 -- Executing [182@default:1] Dial(SIP/181-000a, SIP/182) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/182 -- SIP/182-000b is ringing -- SIP/182-000b is making progress passing it to SIP/181-000a -- SIP/182-000b answered SIP/181-000a -- Remotely bridging SIP/181-000a and SIP/182-000b == Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-000a' Seems like extension 182 (called ext) is getting call and passing them another time to me 181 (origin call) I've try it with siemens pbx and works as expected It's very hard to see what's happening without seeing the SIP logs. You just see that something went wrong in the process of setting up the bridge. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timer1 RFC and SIP.CONF
4 jul 2012 kl. 13:32 skrev Elliot Murdock: Hello, I am trying to get clarity with the sip.conf timer configuration. The current configuration states: ;--- SIP timers ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;timert1=500; Default T1 timer ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 However, according to RFC 3261: (EXCERPT 17.1.1.1) T1 is an estimate of the round-trip time (RTT), and it defaults to 500 ms. Nearly all of the transaction timers described here scale with T1, and changing T1 adjusts their values. The request is not retransmitted over reliable transports. After receiving a 1xx response, any retransmissions cease altogether, and the client waits for further responses. The server transaction can send additional 1xx responses, which are not transmitted reliably by the server transaction. Eventually, the server transaction decides to send a final response. (EXCERPT 13.2.2.4 2xx Responses) The UAC core considers the INVITE transaction completed 64*T1 seconds after the reception of the first 2xx response. According to the RFC, the 64*t1 timeout is not for provisional responses, but for final responses. This seems to be in contradiction to what is stated in the sip.conf file. Unless you have PRACK support, which Asterisk not yet has, there's no timeout in the SIP protocol for provisional responses more than the need to update your provisional response at least every minute not to hit a 3 minute timeout in the SIP transaction state in a proxy. Now, the timerb is used to get ANY response from a server out there, including provisional responses. If we don't get ANY response within TIMERB, the SIP transaction dies and in a UA with a transaction layer we generate a local 408 timeout. In Asterisk, we congest the call. So the 64*T1 is for any response, including final response. It's there to decide whether or not you have intelligent SIP life forms handling your SIP request in the network universe. I hope this clears up your confusion. Regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over SSL TCP or SRTP?
22 jun 2012 kl. 21:59 skrev Bruce B: Thanks. Want to secure everything and anything possible. 1- Can both SIP over TLS and SRTP work in conjunction to each other? Yes. As Kevin said, SIP over TLS only secures the signalling. And it secures it hop-by-hop so every server in the middle can access the content. The signalling should be hidden from other Wifi users, even if it's not hidden all the way between caller and callee. In the signalling you specify how to exchange the actual media. To have secure signalling with TLS doesn't necessarily mean that them media (audio/video/text) is secured. The media is secured with Secure RTP or SRTP, which means that every audio packet is encrypted. 2- Is SIP over TLS a package or added on module that can be installed from Digium Asterisk repository? It's part of the current Asterisk SIP stack, but still marked as experimental as it has a number of known issues that needs to be fixed in order to put this in production use in larger sites and networks. You will have to test it to make sure it works for you. Experimental status means that the configuration options may change in a coming release without being backwards compatible. The TLS part has been experimental in many releases without anyone putting any funding towards fixing it. I guess serious use of TLS is done not with Asterisk but with a SIP proxy like Kamailio or OpenSIPS in front of Asterisk. 3- SRTP takes care of the RTP and makes it secure so that MITM type sniffing is not possible? Yes, provided that the media encryption key exchange is secured. Today, the key exchange is done in SIP messaging, which is why you also want SIP over TLS. Regards, /Olle Regards, On Fri, Jun 22, 2012 at 2:39 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 06/22/2012 12:56 PM, Bruce B wrote: Which one of these ensures that SIP packets are sent and received in a secure format so that users using public wifi don't allow MITM type of attacks or others can't read the plaintext SIP packet info. VPN is not an option. Looking for 2nd most secure to VPN. SIP over TLS (what used to be called SSL) is what secures the SIP signaling. SRTP is for securing media streams. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Community event: Open Source Realtime Dinner in Barcelona - June 13th
Hello! I will be running an Asterisk SIP Masterclass - the last one - in Barcelona in June. During this week, I will organize a dinner for everyone working with or interested in Asterisk, Kamailio and other Open Source platforms for realtime communication. It's June 13th somewhere in Barcelona - location will be announced later. You pay our own dinner (unless we can find sponsors) and enjoy the geeky company for free! To join the event, use this Facebook event https://www.facebook.com/events/307548349321608/ See you in Barcelona! /O -- http://edvina.net - Open Unified Communication - training, consulting, workshops -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Invite + decreasing sequence number = 500 Error?
16 apr 2012 kl. 15:31 skrev Matthew Jordan: It's not a bug - decrementing the CSeq header field value is directly in violation of RFC 3261. From section 22.2: When a UAC resubmits a request with its credentials after receiving a 401 (Unauthorized) or 407 (Proxy Authentication Required) response, it MUST increment the CSeq header field value as it would normally when sending an updated request. This only applies to the same dialog. The question here is if it is the same dialog. If it is, then the server indeed has a bug. Check the Call-ID and the from tag of both requests. /Olle - Original Message - From: Benoit Panizzon benoit.paniz...@imp.ch To: asterisk-users@lists.digium.com Sent: Monday, April 16, 2012 7:12:09 AM Subject: [asterisk-users] Invite + decreasing sequence number = 500 Error? Hi out there We have a strange Problem here with invites. We observe this SIP conversation. C3 PBX - Asterisk Case 1. Sequence Numer always increasing: = Invite = 401 Unauthenticated = Invite+auth with sequence number previous Invite. = 100 Trying etc. Works OK. Case 2. Sequence Number decreasing. = Invite = 401 Unauthenticated = Invite+auth with sequence number previous Invite. = 500 ERROR I was browsing the SIP rfc and I cannot find any clue if in this case the sequence numbers must be increasing (the C3 PBX is wrong) or if I have sumbled over an asterisk bug. Is there anyone who knows? Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- o...@edvina.net - http://edvina.net The final Asterisk SIP Masterclass, June 11-15 in Barcelona, Spain. - Register today! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as register server through OpenSIPS
9 jan 2012 kl. 09:02 skrev Ronald Cepres: Hi all, I've been trying to register a SIP user agent to an Asterisk server using OpenSIPS as SIP router. The functionality is working fine. However, Asterisk uses the IP address of the OpenSIPS server as the peer IP address. How can I use the original IP address of the peer without changing the peer's nat=yes? I appreciate any kind of help. Thanks! You propably have NAT=yes in Asterisk. If you turn that off, Asterisk will save the contact provided by the phone which will point directly to the phone, bypassing the OpenSIPS proxy. In order to get Asterisk to use the OpenSIPS proxy outbound as well you need to define it as an outbound proxy. Now, you have to configure NAT support in OpenSIPS since it's the first hop seen from the phone. /O Twitter @oej Web: http://edvina.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which device auto-registered an extension?
16 dec 2011 kl. 18:12 skrev Barry Miller:
On Fri, Dec 16, 2011 at 05:02:11PM +0100, Olle E. Johansson wrote:
16 dec 2011 kl. 02:03 skrev Barry Miller:
So is there a way for the dialplan to determine which device caused SIP to
auto-register an extension?
Not really, unless someone else can come up with something.
In Asteirsk, the extension hints are the connection from the dialplan to a
device,
used for subscriptions and blinking lamps.
exten = 543,hint,SIP/devabc
then you can use
exten= _5XX,DIAL(/${HINT})
Which opens up the question on how you enter all the hints...
I know Tilghman added something clever recently to new versions of Asterisk,
but I
haven't used it myself so I can't describe how it works.
Thanks. I like to keep devices and extensions separate, for security
reasons, and it's a
pain having to define the association in different places, especially when
sip.conf seems
like the logical place to do this. Do you think if I were to code up a
function like
AUTOREG([context,]exten) that returned 1 for auto-registered extensions, 0
otherwise, and
set a channel variable to the registering device name, it might be useful
enough to be
accepted as a feature?
I would assume that to extend regexten with reghint and have a similar
functionality
where we add a hint to an extension at registration would work nicely within
the
current architecture.
/O
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Re: [asterisk-users] VUC: AstLinux 1.0.0 release
17 dec 2011 kl. 10:11 skrev Darrick Hartman: The AstLinux Team is happy to announce the release of AstLinux 1.0.0. This release includes significant changes and improvements over past releases. Specific upgrade and new installation instructions are available at: http://www.astlinux.org Some of the highlights include: * Using eglibc instead of uClibc. This allows binary compatibility with add-ons that are provided as binary only (G.729 CODEC, Fax for Asterisk etc). * Newer Kernel which better supports newer hardware * Support for Jabber/Gtalk * Removed mISDN support (the zaphfc DAHDI driver is included for single port ISDN cards) A full changelog is available on the release pages. We provide versions with Asterisk 1.8 and 1.4. Because this is a major version change, there are some special considerations when upgrading. Please read the instructions very carefully to ensure no step is skipped. http://doc.astlinux.org/userdoc:upgrade-0.7 I happily noticed support of IPv6! Congratulations to the new release. /O --- http://ipv6friday.org | http://www.voip-forum.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Trunk
16 dec 2011 kl. 11:29 skrev James Courtier-Dutton: Hi, I have a situation where unfortunately, I cannot use IAX for trunks, and need to instead use SIP trunks. Is there any way to fit the voice data from more than one simultaneous phone call into a single IP packet over the SIP trunk. I believe this is possible with IAX trunks, but I don't know how to do it for SIP trunks. Yes there is a proposal for RTP trunking, but it's not implemented in Asterisk. /O http://ipv6friday.org - Learn more about IPv6 every Friday! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which device auto-registered an extension?
16 dec 2011 kl. 02:03 skrev Barry Miller:
Hi all,
In sip.conf:
[general]
regcontext = autoreg
[devabc]
regexten = 543
creates exten= 543,1,Noop(devabc) in context autoreg when devabc
registers. But I can't use exten= _5XX,2,Dial(SIP/${EXTEN}) in the
dialplan, because there's no device SIP/543. Now I know I can add a line
like exten= 543,2,Dial(SIP/devabc) for each and every device that uses
regexten, but it would be a lot cleaner to be able to use something like
Dial(SIP/${WHAT_GOES_HERE?}) instead.
So is there a way for the dialplan to determine which device caused SIP to
auto-register an extension?
Not really, unless someone else can come up with something.
In Asteirsk, the extension hints are the connection from the dialplan to a
device,
used for subscriptions and blinking lamps.
exten = 543,hint,SIP/devabc
then you can use
exten= _5XX,DIAL(/${HINT})
Which opens up the question on how you enter all the hints...
I know Tilghman added something clever recently to new versions of Asterisk,
but I
haven't used it myself so I can't describe how it works.
/O
-
IPv6 Friday again! Woot! http://ipv6friday.org
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[asterisk-users] SIPit 29 in Monaco - interoperability by hard work
Friends, SIPit is an event organized by the SIP Forum and partners. It has been running for 15 years twice a year, making sure that SIP clients and servers interoperate. By testing, we also find issues with the myriad of RFCs in this area and correct them. Testing interoperability is important. The first time I brought Asterisk to SIPit in Stockholm many years ago I was terrified. The SIP stack back then was, well, peculiar. It worked with some SIP phones for basic calls, but not much more. During the tests I learned a lot, got a lot of help from friendly engineers and fixed a large amount of bugs. Afterwards I had a list of todo's that kept me busy for quite some time. It helped Asterisk leaping forward in the SIP area. We've grown since then and Digium, as a member of the SIP Forum, hosted a SIPit in Huntsville a while ago, testing both Asterisk and the new baby, Asterisk SCF. I have hosted a SIPit here in Sweden. The Asterisk eco system believes in interoperability and we work hard to stay interoperable with the world of SIP. This year at SIPit we'll run all kinds of tests. I will personally focus on security and IPv6 tests. Together with the Kamailio/SIP-router team, I've built automatic tests in these areas. Hopefully we can complete them and make them public. (We have a few cool Allison prompts for them too!) I guess I will find a few new bugs as well :-) So why am I writing to the Asterisk-users mailing list about this? It's not about showing off, it's about explaining how interoperability happens. Customers need to require interoperability and open standards, not accept any vendor lock-ins. Vendors and Open Source developers need to take the lead and work together to get interoperability. Customers need to test. So if you are a developer of SIP products that interact with Asterisk - please register for SIPit today. If you are a customer, please ask your vendors if they participate in SIPit. If you have a lot of knowledge about Asterisk, please register and come test Asterisk 10. If you are a happy Asterisk user, please understand that the great level of interoperability that we have in Asterisk, communicating with all kinds of devices and servers out there, just did not happen by accident. It required a lot of work by a large group of developers. And you benefit. I'll try to keep you posted about the progress of the tests. I can't name specific vendors that participate, it's part of the deal. But we can talk about the test results, the state of certain features. Personally I hope that we can succeed in many more tests with TLS and SRTP this event. And we'll have some fun with IPv6 and dual stack implementations as well. Thanks for reading! Have a nice weekend! /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fixing an old bug related to extension s - feedback wanted
Friends, While working with the manager interface, I noticed that an originate action to a non-existing extension had a strange behaviour. Instead of generating an error, which would happen in most VoIP channels and Dahdi, Asterisk started looking for extension s as a fallback. For as long as I've worked with Asterisk, the definition of extension s has been that it is used when *NO EXTENSION* has been given (and in the macro command). There are two good examples - immediate answer in Dahdi and calling a SIP domain without a username part - like sip:digium.com. In my trainings I always repeat (with a loud voice) that extension s is *NOT* a wildcard. Obviously this behaviour is a bug. It's been around for a long time and has been hidden by most apps and channel drivers that handle a bad extension in a correct way and report errors before the PBX is started in order to handle the channel. The question is - how do we fix this? There might be applications out there that depend on this buggy behaviour. What I've proposed are two separate fixes: 1) Change the manager Originate action In Asterisk 1.8, there will be a warning if an extension given doesn't exist, but the behaviour will not change. A flag in Asterisk.conf [compat] section will be implemented so that you can change this behaviour and get an error response in manager if the extension does not exist. In Asterisk 10 the error response will be the default behaviour. If an application using AMI needs a fallback, it needs to be controlled by the application. It needs to know that an extension does not exist and that the call can't be fulfilled. 2) Change the PBX core === The bug actually exists in the PBX core, in ast_pbx_start(). We will not change this in Asterisk 1.8. In Asterisk 10, the core pbx will report that the extension does not exist and no longer fall back to s in current context or s@default. This will, as we see it now, not affect most channel drivers and thus most dialplans. If you rely heavily on the originate function (AMI, CLI and dialplan) and use the fallback behaviour, you will need to modify your dialplans. Final question === My question now is what you think about these changes. Do you need a flag for Asterisk 10 to revert to the old behaviour? Is this bug something you actually rely on in your application? Thanks for your response! /O Edvina SIP Masterclass covering SIP, Asterisk Kamailio - Oxford, UK, Nov 7-11. * http://www.telespeak.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] single registration per user
18 sep 2011 kl. 22:23 skrev Catalin S.: Hello Eric, Is about outgoing calls from multiple devices with the same username at aprox same time. The overwritten is for incomming calls. I want to prevent using the same account in multiple devices at same time. The solution with IP will not apply because users may be behind nat or will change everytime multiple access points. Do you have any other clues? There is no real good way to prevent this. How can Asterisk now which registration that is the valid one? If a device reboots and gets a new IP from DHCP, we do not want to prevent that new registration to prevent the old one from another IP, but the very same device. There's no device ID used in the registration, only the SIP account. This also applies to OpenSER/kamailio/OpenSIPS. We can prevent multiple simultaneous registrations in those, but that will also mean that phones that reboot will be blocked until all registrations expire in the server. /O Edvina SIP Masterclass covering SIP, Asterisk Kamailio - Oxford, UK, Nov 7-11. * http://www.telespeak.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF problem
19 sep 2011 kl. 01:51 skrev Zeeshan A Zakaria: This DTMF problem has always been there and there is no real solution for it, other than using those expensive PBX systems like that from Avaya, Cisco, etc. This problem happens when you are sending inband DTMF tones. Via softphone you are sending out-of-band DTMF which is basically SIP messages. Just to correct the latest part of your statement: The default way to send DTMF in SIP calls is using DTMF as a codec called telephony-event in the RTP stream. This sends DTMF as events. Most hard and soft phones support this - usually called RFC2833 DTMF mode. Asterisk supports it by default. Sending DTMF in the audio usually gets messy when using an IP network. Especially if you use codecs that compress the audio. I do recommend you to use RFC2833. We have built very large IVR services and have no issues with DTMF being received in Asterisk so it's doable. There are other issues with Asterisk DTMF, but that's another issue :-) /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fixing an old bug related to extension s - feedback wanted
20 sep 2011 kl. 15:34 skrev Danny Nicholas: Just my .02 - fix Originate since the Original Asterisk book, page 125 paragraph 1 says s = start. If s is not really start, I'm going to scrap my 3+ years of dialplan writing and change all of my simple dialplans to read exten= start,1,blah instead of exten = s,1,blah. To me exten= s,1,blah is more intuitive and less vulnerable than exten = _X.,1,blah. I am sorry that the Original Asterisk book was wrong and do hope that they corrected that part in later editions. I don't think any official docs have pointed out that s was anything else than a default extension for situations where there is no extension given. Using start makes your dialplans much easier to read :-) and makes them more secure as no app will end up there by accident, which may happen in your current systems. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fixing an old bug related to extension s - feedback wanted
20 sep 2011 kl. 15:34 skrev Danny Nicholas: Just my .02 - fix Originate since the Original Asterisk book, page 125 paragraph 1 says s = start. If s is not really start, I'm going to In the first edition, page 82, it actually says When a call enter a context without a specific destination extension, they are handled automatically by the s extension. Which is correct. It continues (The s stands for start, as most calls start in the s extension) which is very wrong. In the edition you have, page 125, the most calls part is deleted and the text explains that this is where a call will start if no extension information was passed with the call. So they got it right in the end :-) /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overlap SIP dialing
8 sep 2011 kl. 17:17 skrev Kevin P. Fleming: Honestly, I'm not really sure that there is a practical solution here. ISDN overlap dialing was intended for 'dumb' phones, and SIP phones aren't 'dumb' :-) That's a quote that goes to my quote storage layer. /O ;-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overlap SIP dialing
8 sep 2011 kl. 17:26 skrev Andrew Latham: On Thu, Sep 8, 2011 at 11:21 AM, Olle E. Johansson o...@edvina.net wrote: 8 sep 2011 kl. 17:17 skrev Kevin P. Fleming: Honestly, I'm not really sure that there is a practical solution here. ISDN overlap dialing was intended for 'dumb' phones, and SIP phones aren't 'dumb' :-) That's a quote that goes to my quote storage layer. /O ;-) -- I want a t-shirt SIP phones aren't 'dumb' :-) Overlap dialing has very limited use, however I found it helpful when testing integration with other PBX/VM/PSTN connections. Yes, but the solution is not 484, but as Kevin stated to answer the darn call on the SIP side and provide a dial tone from the other side. And yes, we do have dial tones on ISDN PRI trunks... /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set(CHANNEL(musicclass)=
6 sep 2011 kl. 22:30 skrev Leif Madsen: On 02/09/11 11:27 PM, Joseph wrote: In asterisk 1.4 I had: exten = s,n,Answer() exten = s,n,SetMusicOnHold(default) But in 1.6 1.8 I think don't need to use: SetMusicOnHold(default) (beside it is deprecated) as it is default. In 1.6 and UP I think it is: Set(CHANNEL(musicclass)= Can somebody explain what do they mean by CHANNEL? CHANNEL() is a dialplan function. You're setting parameters for the current channel by using that function. So instead of using a dialplan application like you were before, you use the CHANNEL() function. exten = s,1,NoOp() same = n,Set(CHANNEL(musicclass)=default) I could use just: exten = s,n,MusicOnHold() There is a lot of documentation on www.voip-info.org but sometimes it is not clear which asterisk version it applies to :-/ (Another good reason to be reading the documentation on https://wiki.asterisk.org/wiki instead :)) ...or update and help maintaining the voip-info documentation. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overlap SIP dialing
7 sep 2011 kl. 15:59 skrev Daniel Tryba: Looking at the history of the list I don't expect any answer but lets try anyway: Does anybody use overlap dialing from SIP devices to asterisk? Does anybody have a working example? To add to your question: Does anyone have a phone that supports this properly? /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overlap SIP dialing
7 sep 2011 kl. 16:20 skrev Andrew Latham: On Wednesday, September 7, 2011, Olle E. Johansson wrote: 7 sep 2011 kl. 15:59 skrev Daniel Tryba: Looking at the history of the list I don't expect any answer but lets try anyway: Does anybody use overlap dialing from SIP devices to asterisk? Does anybody have a working example? To add to your question: Does anyone have a phone that supports this properly? /O Yup, I have a few... http://wiki.snom.com/Settings/overlap_dialing Great. Haven't seen this - thank you. The whole concept is interesting. Suppose the call forks and one UA answers with 484, another with 486 and another with 180 ringing. What are you supposed to do? I think there's a problem with the RFC 3261 here and don't know if it's been clarified. Now - in the case of Asterisk if we call out to two devices from the dialplan and one responds with 484 and another with 180 ringing - what happens in Asterisk? /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted a modified SIP message body
31 aug 2011 kl. 14:42 skrev Kevin P. Fleming: On 08/31/2011 02:46 AM, Jaime Lozano wrote: Hello, I agree with you, I'm not explaining the problem in a proper manner, because of my lack of Asterisk knowings. I send the Wireshark captures. 3com telephones take the timezone TZ:7200 from the 3Com PBX to show the time right. But what if I want a 3Com telephone to work with Asterisk PBX? Then the telephone time is wrong, 2 hours lower. It seems 3Com telephones need the TZ:7200. 3Com telephones work with Asterisk and it is great, but we would like to log the calls. OK, so the first clarification is that you are talking about responses to REGISTER requests specifically, not all responses to all requests. That's good :-) On to the meat of the issue... indeed, the '200 OK' response to a REGISTER request does not normally have a message body; nothing in the SIP RFCs even suggests that there would be one (although it's certainly allowed should the registrar want to include it) or what would be present in it. As has been previously replied here, there is no facility in Asterisk to include a message body in a REGISTER request response, so providing one will definitely require source code modifications. They wouldn't be terribly difficult, but they would only be applicable to these particular phones, which reduces the benefit of making the changes to the community at large. With that said... it's certainly possible to do this, but it's going to take some non-trivial code changes. The REGISTER handling code does not use any of the methods that exist in chan_sip to add message body content to its responses, it uses simpler methods that assume there won't be a message body. In addition, this mechanism is really pretty broken anyway; the server would have to know where each phone is physically located in order to be able to provide the correct TZ value to it, and would have to be updated if the phone is moved. Not an ideal situation. The RFC states that a phone could use the Date: header in the response to set the local time in the device. It's always in GMT which makes it stupid to add a time zone any where. -1 for this implementation. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] seeding an originated number in a SIP phone [was: Re: Thunderbird extension using AMI to dial]
29 aug 2011 kl. 15:05 skrev Kevin P. Fleming: On 08/28/2011 01:56 AM, Tzafrir Cohen wrote: On Thu, Aug 25, 2011 at 07:36:53PM +0100, Chris Hastie wrote: Hi I've just added direct support for AMI to a forthcoming version of TBDialOut, a Thunderbird extension for dialling direct from Thunderbird's address book. If anyone fancies testing it I'd be grateful for any feedback. If you feel like casting a critical eye over the code, or doing some translating, even better. AMI support is available in TBDialOut 1.7.0pre1, which can be found either at http://www.oak-wood.co.uk/tbdialout/ or from the 'Development channel' at the bottom of the page at https://addons.mozilla.org/en-US/thunderbird/addon/tbdialout/ We already have a dialer script (sent to this list a while ago) so it's good to see that this extension support that simpler option as well (I don't use ThunderBird, as you can see. Some others in the office do use it). One followup question: I originate a call from a SIP phone to some remote number. The problem is that the number will not show up properly in the list of outgoing calls for the phone. Any idea how to fix this (for whatever SIP phone)? You aren't originating a call *from* the phone (that would require some sort of API into the phone itself to make it place a call). You are originating a call *to* the phone and also to another endpoint; as far as the SIP phone is concerned, this is an incoming call. I've never seen discussion of a desire to provide a method for an incoming call to be treated as if the endpoint had placed the call itself in any of the SIP discussion lists I frequent... so I'm pretty sure there's no standard way to do this. Oh, there is - REFER. We could possibly implement sending a REFER request to the phone, something that is frequently used to do call setups from click-to-call apps. This is not something we do support in Asterisk today. I've implemented it using SIP libraries since Asterisk doesn't have to be involved in the REFER. If you do ORIGINATE from the phone you have to be aware that Asterisk lacks some security framework here. An application that has ORIGINATE access can reach the whole dialplan. I have patches for that which needs to be moved forward. My proposal is to add a default context to manager accounts to put a limitation of destinations they can reach with originate and redirect AMI commands (which where the only ones I could come up with as dangerous in this regard). /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted a modified SIP message body
26 aug 2011 kl. 14:06 skrev Jaime Lozano: Hello, In which file do I use SIPAddHeader()? Please consider that the packet goes from the PBX to the telephone, and what I want is not a header because the TZ: 7200\n is in the *message body* not in the *message header*. That's no longer a SIP header, it's part of the SDP you want to change. You can't do that without changing the source code. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem setting for incoming termination
12 aug 2011 kl. 14:51 skrev Kevin P. Fleming:
On 08/11/2011 02:03 AM, Jim Boykin wrote:
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is that whenever call
is received from the client, it goes to default context instead of
'dallas' context. Also, the ${CDR(accountcode)} variable remains
empty. Now, If we set fromuser field at the client end, then
everything starts working, however, in that case, it overrides the
callerid.
This is a known and well-understood problem caused by the method that
Asterisk users for SIP authentication; the 'From' header in the incoming
INVITE is used *both* for determining which user is placing the call and for
Caller ID. As you note, if you have the real Caller ID in that header, then
Asterisk can't use it for matching to a user in sip.conf, and thus can't
authenticate the call properly.
The solution for this is to use 'sendrpid' on the sending end and 'trustrpid'
on the receiving end; this will configure Asterisk to transfer the Caller ID
information in a Remote-Party-ID (or P-Asserted-Identity, depending on the
version you are using) header, allowing the From header to be used solely for
authentication.
Or stop using type=user and type=friend, and stick to type=peer and ASterisk
will only match on IP+port address.
/O
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[asterisk-users] Asterisk/Kamailio dinner in Madrid thursday next week - June 30th
Next week I'll be in the hot city of Madrid doing Asterisk/Kamailio training - The SIP master class. Maybe we can organize a voip nerd dinner on Thursday evening? If you're interested, please e-mail me off list and I'll send out more details later. Greetings /Olle PS. E-mail off list :-)-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
31 maj 2011 kl. 14.49 skrev Benny Amorsen: Jeff LaCoursiere j...@sunfone.com writes: Hasn't anyone managed to solve this with something better than a caching DNS server, which seems to only last a short while? What exactly is going on that is failing? If your recursive DNS server returns errors quickly rather than actually trying to look up the names, Asterisk works fine. It is not a particularly nice workaround, but it does work... As long as Asterisk does not actually NEED the DNS information, but that can be most worked around with static configuration of IP addresses in sip.conf. Longterm we should really integrate an Asynchronus DNS library, like C-Ares. I've been wanting to do that for years. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending call to specific IP address
23 maj 2011 kl. 23.36 skrev Paul Belanger: On 11-05-23 05:30 PM, Elliot Murdock wrote: Hello, I am wondering how to send a call to a specific IP address that is different than the host of the URI. For example, an invite to the URI is j...@phone.com needs to be sent to the IP address 123.456.789.255, not to the IP address of phone.com. How is this done? Look at the 'Contact' header. I don't what Paul means here... YOu can surely define a peer and add an outbound proxy with the IP address... That way we won't overwrite the domain. I am not aware of a way of doing it in the dial string in the dial plan. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP per-call heartbeat?
24 maj 2011 kl. 12.19 skrev Tony Mountifield: One of our customers has an Asterisk conference bridge connected to a SIP trunk from an ITSP. Yesterday, they had two inbound calls that didn't get hung up properly. From the tcpdump SIP trace that we have running continuously, I can see that no BYE was received by the bridge, and when some hours later the hangup was forced from the bridge end, the bridge sent a BYE to which it received a 481 Call Leg/Transaction Does Not Exist. If the remote end send a BYE and doesn't receive a response, that bye will have to be retransmitted multiple times before it gives up. The SIP protocol includes retransmission over UDP, to cover up for packet loss. If it did not retransmit, you have other issues. Since SIP is UDP, this situation must occur from time to time, and I wondered if it is possible to configure any kind of per-call SIP heartbeat so that a dead call could automatically be identified with a 481 response much sooner. SIP session timers is what you need for that. Implemented in Asterisk 1.8. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
5 maj 2011 kl. 18.30 skrev Ira: At 07:56 AM 5/5/2011, you wrote: So how can we fix this? How can we get more people involded? What makes projects like FedoraTesting[3] and DebianTesting[4] popular? How can the Asterisk project reproduce their success? Well, it's not a lot of people willing to run beta software on their phone system. Phones need to work and for most people they need to work perfectly all the time. I'm one of those oddities that will always run beta software if given the chance but my experience is that quite rare. As I've said before, I'm more then willing to help with answering questions about the testsuite or reviewing code that people want to get merged in. We also have an IRC channel, #asterisk-testing available for people to join, ask question, idle, lurk, etc, or if you want to reply to this thread, feel free. But get involved! :) So I'm the person who has never been able to keep 1.8 alive on my system for more than a minute or two and I've probably tried more than 10 different betas and release versions. I posted a bug report which was closed in minutes, I posted the problem on this list every few tries and zero response. I tried to figure out mIRC. It's installed on my machine but I've never got past that. I just don't get the instructions. I know that all the people involved in the project are Linux heads, but some of us, like me, have a Linux box only because of Asterisk and if you want my help, you need to make being involved accessible and stop assuming we all know what you know. I see the words, jut post a bug report on Mantis posted all the time and I'm sure it means as little to others as it means to me. Maybe there needs to be a web page somewhere, Asterisk beta testing for dummies so that you can point us to so you don't have to answer the stupid questions over and over. I've beta tested enough and had enough beta testers to understand the kinds of things that make it possible to get bugs fixed, but it's usually a very small percentage of users that understand that. Thanks for the feedback, Ira. It makes me very sad to hear what you say and I hope that we can get more resources from the community to assist in the process to make it more friendly. We want to get those bug reports. The one thing I hate to hear when I'm travelling at conferences is that oh, I known that bug for a long time but did not bother to report it. Apologies for your experience with the bug process. Regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
5 maj 2011 kl. 05.28 skrev Flavio Goncalves: My 2 cents. All these problems seem to be lack of focus. Digium, please stop doing everything to everyone. Too many versions, too many features, too many code, too many bugs. Following the Pareto's principle, 80% of the users use only 20% of the code. My suggestion is to start thinking of Asterisk as a platform taking care of only 20% of the code. Digium is in position to create a market place for free and commercial Asterisk applications, drivers and modules. Look at some other open source communities such as Joomla at http://extensions.joomla.org, There are more than a thousand modules maintained by the community. Imagine, do you want a multitenant parking module? Great there is one in Digium App Store for a few dollars. Digium could have its own commercial modules. Support for 3rd party applications would be up to the 3rd party developers. Why iPhone developers make money and Asterisk developer's usually don't? If people pay for silly games in iPhones wouldn't they pay for a Unistim driver if they have hundreds of compatible phones? What you are describing is in the architecture for Asterisk SCF. The F stands for Framework, which hints at the ideas behind it. Asterisk as it exists today has been around for a long time. There are many, many extensions around and we also have the forge and the marketplace. It will be hard removing stuff from the distribution, imagine the reaction - considering my latest reaction to the 1.4 actions :-) I would like to say that I have a deep respect for Asterisk and Digium that redefined the global telephony market, but stuffing Asterisk with many new features on each version does not seem to be contributing to the stability of the code or the migration to newer versions. Thanks for your input! /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
5 maj 2011 kl. 06.33 skrev Olivier: 2011/5/5 Flavio Goncalves fla...@asteriskguide.com snip but stuffing Asterisk with many new features on each version does not seem to be contributing to the stability of the code or the migration to newer versions. yes but it seems to me that code stability is improving. Maybe next 1.10.0 version will be production-ready from day 1 ? Unless a lot of users step in to test the pre-releases, that will not happen with new code in this project or any other project. It just takes time. The more people that test, file bug reports, patch code and helps us through the process, the better. I would like to suggest that the community put more eyes towards helping with the test system. The test system, as Russell pointed out earlier, is a huge improvement that saves us from repeating a lot of mistakes. Ideally, for every bug fix we should add a test to make sure it doesn't come back... The test system makes every release better than previous releases. And it can help in your installation as well! /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
5 maj 2011 kl. 12.04 skrev Paul Hayes: On 05/05/11 05:41, Cary Fitch wrote: Flavio E. Goncalves www.asteriskguide.com http://www.asteriskguide.com Compare to which version of Windows… Patches are a never ending process Cary Fitch I think this attitude is half the problem. Asterisk is not a desktop computer operating system. It is the engine for a telephone system, a telephone system needs to be much more reliable than a desktop PC if it is going to continue to compete in a growing industry. I agree with the comments on concentrating more on stability than new features. It's hard because it is new features that make good stories and are easier to shout about in order to get a product better known. For now I am sticking with 1.4 mainly (although I am using 1.6 where I need BRI connectivity) but my plan is to move to 1.8 when I feel I have tested it enough and it's been around for long enough to be proven. Great. You are part of the test team :-) One has to remember that this is open source. We need to work together to stabilize 1.8. No one else is going to do it for it - and I feel it needs to be done. Thanks for your help! /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP secruity: username and password
5 maj 2011 kl. 14.08 skrev bilal ghayyad: Hi All; When the endpoint register on Asterisk or initiate a call, so they exchange the sip username and password. What is the possibility that this will be capture by the hacker and how to avoid this problem? We never exchange passwords in clear text in SIP 2.0. SIP uses HTTP digest authentication with MD5. There are many articles about that on the web, so that you can find out how it works and what the risks are. Cheers, /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP secruity: username and password
5 maj 2011 kl. 14.17 skrev Alex Balashov: Bilal, On 05/05/2011 08:08 AM, bilal ghayyad wrote: When the endpoint register on Asterisk or initiate a call, so they exchange the sip username and password. What is the possibility that this will be capture by the hacker and how to avoid this problem? Strictly speaking, there is no inherent connection between either registration or call initiation on the one hand, and authentication. Both of those scenarios can be performed in an authentication-free fashion. In fact, in most cases the SIP UAC will first attempt to send both a REGISTER and an INVITE request without any authentication credentials. Because they HAVE TO. In the 401/407 reply, there's a challenge (nonce) which is an important part of the MD5 Digest Auth scheme. However, it is typical of a SIP UAS providing retail services to the public at large to reply to those requests with a 401 or 407 proxy challenge requesting authentication. The UAC then resends the request with digest authentication headers, including a password encrypted via a cryptographic one-way hash function. The entire mechanism was borrowed from HTTP digest authentication. The authorisation username can absolutely be intercepted, as it is transmitted it in plain text. But this is not news. The password is encrypted, and while the encrypted version can be intercepted, it is encrypted using a one-time nonce value that is part of the 401 or 407 challenge sent by the UAS. Nonce values typically have fairly stringent expiration times, at least on good implementations, but nonce replay attacks are possible in principle. The password is NOT encrypted. It's is used as the basis of a textstring you calculate a hash from. That's very different :-) This mechanism is reasonably secure, as a compromise with the interoperability requirements of providing SIP service across the public Internet. In high-stakes situations, however, it may not be sufficient, and may call for SIP over a TLS transport, or encrypted tunnels. I would say it may call for SIP with TLS client authentication - regardless if you need encryption or not... Cheers, /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory
5 maj 2011 kl. 15.11 skrev Paul Hayes: On 05/05/11 14:04, Jonas Kellens wrote: Hello list, what does this mean : [May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause code, buddy. The cause code!!! [May 5 14:58:12] DEBUG[8770] chan_sip.c: This call was answered [snip] see rfc3326 section 3.1. Call Completed Elsewhere. It's used so that phones in ring/hunt groups don't record a missed call if the call is answered by someone else. I was looking forward to Asterisk supporting this for a while :) We've had that for quite some time. There's an option to Dial() and one for Queue() to enable it. Check the documentation. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Orginate not working well with PSTN lines
5 maj 2011 kl. 16.35 skrev Gilles: On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell sruff...@digium.com wrote: I know this thread is dead but: I do not believe this should go into the DAHDI kernel modules. I agree. It's just too bad Dahdi is unable to report how an outgoing call is doing: Still ringing, busy, answered. Just to add to the confusion... I have a branch where I managed to get manager originate to handle early media. If we get 183 (sip) or progress in ISDN with media before the answer, a manager originate will start the bridge. We're using that to get the Telco messages when we dial out to connect to a meetme. Previously we just had failed calls, but now we can hear the Telco message saying something like Invalid number, please try again or Weasles have eaten your phone system In the SIP channel, I would like to send some sort of control message when we get 100 trying. This means that we at least have a connection to something, even if we don't know if we've reached the target endpoint. I don't know if there's a similar message in ISDN, PSTN or other channels. But that's another patch :-) /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Test platform
Here is the thing, there is nothing stopping 'the community' today from doing this. In fact, we already have a testsuite [1] in place, running each subversion commit and producing results for the last year. But this is only one type of testing; automated, we also have unit tests built into Asterisk that run too (EG: a unit test to parse SIP URI). Again, each subversion commit we run the tests and validate results. I think we should make it more clear and give examples on how we can extend the test platform to test functionality in our own platforms - our dialplans and channel drivers. If we did that, more people would use the test toolkit and work with it daily. There is still lots of work that needs to be done though. More test plans and test cases to be added, more code to be written and libraries added, getting more people involved in testing Asterisk Release Candidates (RCs) or patches on the issue tracker. That is the hardest part, getting people involved. Sure it is easy to say Asterisk is not stable, not production ready or it crashes all the time; fair enough but we have tools in place to help resolve that. Just in this thread alone I don't believe one person has answered the call of Olle to volunteer time to help maintain Asterisk 1.4 (if I am incorrect please speak up, I must have missed your name). Additionally, this almost exact point was raise on the asterisk-dev mailing list in 2009 [1] (a great read BTW, lots of great ideas) however due to the lack of interest it did not go to far. If you go even further back, Russell and I had a branch where we started some early work many, many years ago. We're asterisk-dinosaurs in that respect... I am very happy that we now, eons later, have a test toolkit. It's lightyears ahead of what we discussed or dreamed of back then. And it has helped a lot in catching stuff. So how can we fix this? How can we get more people involded? What makes projects like FedoraTesting[3] and DebianTesting[4] popular? How can the Asterisk project reproduce their success? Give them something that tests their own setup as well as test the Asterisk in the core. As I've said before, I'm more then willing to help with answering questions about the testsuite or reviewing code that people want to get merged in. We also have an IRC channel, #asterisk-testing available for people to join, ask question, idle, lurk, etc, or if you want to reply to this thread, feel free. But get involved! :) Absolutely - we need people that test the new bugs that developers invent :-) /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Password to be ecrypted?
4 maj 2011 kl. 19.44 skrev Robles Román, José Miguel: By the way, I like the implementation in iax.conf (auth=md5 ... secret=x), it seems more flexible, and it enables the use of other hash functions or other security algorithms. The SIP protocol does not support any other hash functions today. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?
2 maj 2011 kl. 18.09 skrev Hans Witvliet: On Thu, 2011-04-28 at 11:25 -0400, Bruce B wrote: Hi everyone, How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is there any easy way to simulate a distorted SIP line temporarily for testing? You can intruduce a predefined amount of distortion on your ip-connection (packet loss, fluctuating delay, out of secuence reception of packets, limited bandwith) All of these will have a serious impact on your VOIP-connection. See lartc about it. Good thing about it, is that you pre-define how bad a line is, and it produces re-producable results I use a laptop with a usb-ethernet connected in bridge mode as a voip destroyer. Using TC you can inject a lot of bad stuff on the connection. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
I don't think there's anything inherently wrong with the bug tracking system. It's more of a resource issue with many conflicting priorities. Officially letting off some of the pressure from older branches does help. I would like to be making faster progress through bug reports and patches. I do have an open position for another full time Asterisk developer at Digium in case anyone is interested. :-) I agree with Russell here, we have resource issues in the bug tracker but that's nothing that can be solved by another piece of software. If you have issues that is not handled timely, why don't you spend some time with other issues to help out? Surely there are issues where you can give a helping hand. In answer to an earlier email that I felt was kind of attacking me I would like to point out that I am very happy and grateful about the resources that Digium put in the project, and continue to do. Just to clarify that this discussion was not about trying to paint Digium as a company as evil - which I was accused of. Digium is a very old business partner to my company and we've done great together. That doesn't mean we can't critizise each other or not want to discuss issues in the open. To answer another attack, I have been contributing code and bug fixes to both 1.8 and trunk. Most of my code exist in versions for trunk and 1.4. Customers pay me for 1.4, I forward port it to trunk when I have time and resources over. It's not a personal choice that most of my development work still is based on 1.4. Of course I would love being doing development freely, creating great new code for the new release. There's a lot of stuff to do in Asterisk trunk, but no one out there that wants to put resources towards it in my direction. Asterisk trunk development is sadly too far away from my customers current business. The 1.6.x release schedule widened that gap and we need to discuss how to close the gap again. We do not need a large number of maintained releases between the long term support releases. So far I haven't seen more than a few people that chimes in to this discussion saying we need to have 1.4 open, I haven't seen many people running 1.8 in production either. I have seen a lot of important issues being reported with 1.8 which to me confirms that it's still not ready. I have been working in commercial software companies for a long period in my life. A product manager that called for end-of-life of the 1.4 release at this stage would be out of a job very soon. Migrating a customer base from one version to another is very, very hard. It seems much harder in telecom software than in the rest of the software world. We need to continue to work on 1.8 and do a lot of marketing for upgrading as soon as we're comfortable with it and have resources to manage the bug reports that will come in. We really need to push and shove. What I can't do to my customers is forcing them to upgrade to something that doesn't work. Customer will simply stop paying me if I do. I will not continue to push this issue, just realize that I will have to manage my own 1.4 branch fixing the issues that affect my customers, which will exclude management of a lot of modules that are not used at all in our installations. As I said before, I have no resources to support all of the code base for everyone. That's just life, painful as it is. In the ideal world, there would be resources to help everyone. Unfortunately, I still have to have money to bring home at the end of the day. Thanks for a very good discussion. As usual, I learned a lot from it. Keep reporting issues so that all of us can move forward to new releases. Feel free to contact me off-list if you want to discuss this further. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
29 apr 2011 kl. 01.49 skrev Leif Madsen: Well the issue is that we currently have over 900 open issues in the Asterisk project alone, and with only one primary bug marshal (myself) sometimes things accidentally get closed if it looks like a configuration issue. What's the reason that we only have one bug marshal? We used to ask people to become bug marshals to help, but the last I heard you and Russell did not want community marshals. What went wrong with that? Wasn't it any help? /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: 1.8 quality issues
28 apr 2011 kl. 16.53 skrev Russell Bryant: - Original Message - PS. Please don't start a discussion about 1.8 quality in this thread, that's a separate issue. I just want to know what you think about closing 1.4 support now. If you want to discuss 1.8 quality, start a new thread. Thanks. I don't think it's a separate issue at all. I would like to see discussion of exactly which issues are preventing users from using Asterisk 1.8. We're trying to shift focus to those issues and get them resolved as quickly and as efficiently as we can so that we can all move forward. Thanks for ignoring my plea... Please at least change the subject ;-) Resources are limited. What is the best use of our time to help ensure the best future? Where do we want to see the project in the next 6 months to a year? A primary focus on further solidifying Asterisk 1.8 is what gets us there in my mind. I agree. Asterisk 1.4 was released 4.5 years ago. It mostly just works, and I fully expect many to keep using it until they see a need to migrate. If you think it's mostly just works it can't be hard to support it a while longer then, can it? This process has been likened to when the community moved from Asterisk 1.2 to 1.4. Asterisk 1.8 has been much more stable out of the gate than 1.4, due to many things we have done over the years to increase quality, including: 1) We have adopted peer code reviews as common practice for all non-trivial changes going into Asterisk. This alone has _greatly_ increased the quality of the code going in. It is rare that a patch goes up for review where someone doesn't point out some sort of problem. These problems are found and fixed _much_ faster in the up front review process than if it had been many months later when someone encountered it as a bug in the field. Agree. But it also puts a significant delay on the process. We have to be very careful about that. Having too many branches open in addition to this was a pain. With fewer branches I hope it will get better. 2) We have placed an increased emphasis on automated testing efforts. In addition to building up a lot of test environments inside of Digium, there is now an open source automated testing effort for Asterisk. There are over 200 test cases that run every time anyone touches the code. This includes complex call scenarios such as transfers and call parking. These open source test cases touch about 25% of the code (and what it does touch are things we considered some of the most important parts). That is a huge step forward from where we started. We are continuing to place more and more resources on this effort to move it forward. Agree. It's great and we need to continue working on it, because it obviously hasn't caught everything we should have caught. I fully agree that it is a wonderful system and I've said that many, many times. Despite comments in this thread, there _are_ many people using Asterisk 1.8 in production, including large installations. The ones with systems working perfectly fine don't tend to make as much noise. :-) For those still getting hit by problems, I hope that you can make the time to report them so that we can work with you to get them resolved. I don't disagree there either. I have only stated that it fails in my and my customer's installations. Everyone is using Asterisk in different ways. If it did not work anywhere I would be very disappointed. I started my work on Asterisk as a volunteer 7 years ago and even though it is now my full time job, I still put many personal hours into the project. I care very deeply about the success of Asterisk. I truly believe that the steps we have taken with release management are in the best interest of the project. I understand that you do, I don't think you do things you don't believe in. But you do need feedback from production sites to make the best decisions. What you bring up here is important but in my world have no relation to the decision about 1.4. I understand you want to use development resources in a good way, but there are also marketing/business perspectives to consider here. I personally don't think closing 1.4 support today is in the best interest of the project from a marketing point of view, as I don't believe we have a working alternative to offer. I understand we have different opinions about it. Regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Discussion: Are we ready to leave 1.4 behind?
Friends, We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. According to the release plans, support for 1.4 was scheduled to close in April 2011 - basically now. After that, only security patches would be committed. This is already a delay from the original plan published by Russell Bryant. Unfortunately, I think this is way too early. My feeling and experience is that 1.8 is not ready for production in the environments I work in - large scale installations. Customers are not planning migration and all new installs are still 1.4. Tests we've been doing with 1.8 has failed within just a short time and so badly that customers has not paid me to spend any further time with 1.8. Last time we went through this process with a LTS release (which we did not know then) it took over one year before we had a stable product to migrate away from 1.2 and jump on the 1.4 track. Hopefully, with the help of community, we can move up to 1.8 late this year or early next year. For me 1.8 is the focus, it's the LTS release. Not having a supported 1.4 version from the Digium-hosted repositories will mean that we will have to move to separate repositories or branch off from the main track. I already maintain a ton of subversion branches with various patches to 1.4 It takes a lot of time to manage this version that is a fork from the main 1.4 branch. I will soon have to start working with subversion branches for 1.8 to create a compatible version for my customers to test, since most of the patches is not part of 1.8. After a few years of doing this, I know the work involved with managing code myself. The Digium team wants to go ahead and not support 1.4 any more, I want to keep 1.4 open for normal bug fixes. What do you think? Kevin proposed that the community maintains the 1.4 branch without support from the Digium team. I don't think that's a good solution, but it may be the only solution. I haven't got the resources to manage the 1.4 code myself, so I won't step forward as a maintainer if I can't get proper funding. Anyone else out there that has the time and resources to manage the code? Feel free to send me mail off list if you have ideas or suggestions on how to solve this - or continue the discussion here. Regards, /Olle PS. Please don't start a discussion about 1.8 quality in this thread, that's a separate issue. I just want to know what you think about closing 1.4 support now. If you want to discuss 1.8 quality, start a new thread. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
I(me, my opinion, my feelings, my commercial view) am on the side of dropping support for 1.4 and 1.6. 1.8 had some major issues which are resolved/being worked on with more energy as older platforms are shut down. If a large enough security issue showed up, I hope we would all try to do the right thing and push it back to 1.6 and 1.4. 1.6.x is not an option for me at all. These' releases are not LTS. We can't upgrade as often as that release schedule required. I am very happy to see 1.6.x disappear in the darkness and from my hard disk drives. Support must end sometime. Merging changes across many versions is very difficult and time consuming. I fully agree here. Asterisk GUI is very limited do to its 1.4 support code. There are users that still use 1.2 and are very happy. They are not looking for new features. I hope the 1.4 / 1.6 users can survive while they test the 1.8 branch and share why or why not it will fit their needs. They will survive and they will merge their own bug fixes. I just wish we could share the work and maintain the branch in public instead of everyone managing it by their own. As long as 1.8 is not ready for the way we use it, we have no version to migrate to. I am sure that 1.8 will fit their needs and deliver a lot of extra. It's a cool new release. Everyone wants to go there. That's not the issue here. The issue is when it's ready for the larger installed base beyond the early adoptors. I don't like the project I've been part of for many years not offering a supported option that fits the customers I work with. It's as simple as that. Saying that they should know better, that the project has posted the release plans for a long time warning about this - it just doesn't cut it as long as we have no working code to replace the current version with. Compared with last time we had a painful migration (from 1.2 to 1.4) there are numerous other options out there. I think the project have to be a bit more careful about our attitude towards the installed base. I want to keep them in the Asterisk project. That is where I belong and where they belong. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The SIP channel driver - I'm giving up.
Friends, After having spent many years working with the Asterisk SIP channel driver and the SIPv2 protocol, I have finally realized that this is a dead end. It's getting nowhere and it's way too complicated to set up, run and support in working code. After realizing this, I started a new standardization project together with my friends in Canada, Simon and Marc, to develop a working solution based on the combination of IPv6 and SIP. We have gotten great feedback and now the IETF, the ITU and the IPv6 forum jointly launches the new standard, SIP-six. From the press release: ”We realize that 99% of the SIP users use SIP for PSTN phone calls. The original SIP standards was written with other applications in mind, a vision that never came true.” said Bob Plug, area director in the IETF. ”That’s why we sat down and said to ourselves that this should be way more simple.” The SIP-six standard totally removes the dependency of domains and URI syntax. There’s no point in using this, since everyone seems to think that IP addressing is more than enough. The new standard use part of the vast IPv6 address space to incorporate the E.164 phone numbers as addresses. This is the reverse of the reverse phone number usage in the enum standard, which is no longer needed in SIP-six. By using IPv6 mobile IP, phone users register their phones and get access to their phone number. Users that need security can easily integrate IPsec into their setup. Media and signalling uses the same addressing scheme and is mixed so that both SIP-six, RTP and RTCP only uses one port address - but in this case a port address with 32 bit subaddress identifying the media stream. This finally solves a lot of the firewall traversal issues that SIP v2.0 had. With the combination of mobile IP and use of public IPv6 addresses NAT traversal won’t be an issue. The testbed for SIP-six has been running for a year at six choosen large SIP carriers, with the assistance of Edvina AB in Sweden and ViaGenius in Montreal, Canada. In an International effort, the testbed is today put in production and Roboid phones all over the world is automatically connected to this worldwide network. You will be able to find out more about it here: http://bit.ly/sipsix SIP-six is implemented as a channel driver in Asterisk 2.0, as a replacement for SIP2.0 in Kamailio 4.0 and a channel module in FreeSwitch - all releases to be released later today. Softphones for testing will shortly be available from Blink and Zoiper. Looking forward to continue this project with the rest of the Asterisk community! Have a nice weekend! /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP endpoint registrations
16 mar 2011 kl. 14.13 skrev Benny Amorsen: Kevin P. Fleming kpflem...@digium.com writes: Why do you need a Local channel to do this? If extension 234 exists in some context, the Dial() statement in that extension can dial SIP/234-foo and SIP/234-bar itself. Good point. It can be a bit of fun keeping track of the phones when they are added to or removed from queues, and the owner expects both of them to be added/removed at the same time. It is still doable without Local channels. Once you need to do manipulation of calls before passing them on to the phone (change callerid individually, handle tT options etc.), Local is unavoidable, but at that point multiple registrations would not work either. We also need to adopt SIP headers depending on device - so we need a dial per device, not one combined. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime SIP, multiple AX servers question
3 jan 2011 kl. 00.26 skrev Bryan Field-Elliot: Normally, no matter which Asterisk server an ATA connects to, we get our database fields filled out correctly, such as regseconds, lastms, ipadr, etc. However, with some ATA's we are experiencing a problem as follows: 1. ATA reaches its re-registration timeout, which we typically configure to be 60 minutes. 2. ATA re-queries DNS SRV record, and ends up re-registering with a different AX server than it was on previously. 3. The new AX server updates the Realtime DB fields (regseconds, lastms, etc). 4. The old AX server, after a few more minutes, notices that the ATA has vanished, and therefore clears out these same fields. Oh, that's an interesting observation. Depending on how you see it, it's a bug or a feature request. Code-wise what you could do is that Asterisk could retrieve the information from realtime. If the sysname is not the same as the systems, it let the information be. If it's the same sysname, then erase the information. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing sip port
11 nov 2010 kl. 23.25 skrev Baha @ SH: Hello How can I run the sip service on asterisk on another port beside 5080? I mean asterisk will still take sip requests on port:5080 and another custom port, lets say port:6080 For UDP, we only have one port. You have to select. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: certificate for softphone
10 nov 2010 kl. 21.48 skrev Hans Witvliet: On Wed, 2010-11-10 at 08:38 +0100, Olle E. Johansson wrote: 6 nov 2010 kl. 15.30 skrev Hans Witvliet: Hi all, As stated in the subject, slightly off-topic, as it is not directly a Asterisk issue, but more SIP in general Because security in general, and specifically identification becomes more and more a subject for more concern, and Asterisk is capable of doing sip/TLS, i was wondering what more could be done to improve security. Specially softphones, might it be possible to employ etokens or smartcards for holding the certificates needed by TLS? Done before? In the SIP protocol there is support for TLS client certificates, much like in HTTP. Asterisk doesn't support it. You need to put a SIP proxy like Kamailio in front of Asterisk to get this kind of strong authentication. /O Am i that mistaken? I got the impression** that sip-registration of a phone could be done in the same way as client-authentication on apache: On the server-side you got the certificate holding your public key which is signed by a trusted third party (the CA), while you hold your private key on a smartcard or token. If you start your browser you are prompted for your pin-code. I was just hoping that there would be a softphone that could work the same way, two-factor authentication. I haven't seen any soft clients implementing this. Bria/Eyebeam may have it, but they've removed all TLS options from the GUI. As I said, the SIP protocol supports it. Kamailio supports it on the server side. Now we need clients that supports it. Now we're talking about authentication. For identity assurance, there's another set of standards called SIP Identity where you use TLS to sign your identity. The TLS is just between the phone and the first server. Identity is supposed to be something that follows the call to the callee. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
10 nov 2010 kl. 02.38 skrev Brett Woollum:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2]
NoOp(SIP/413-0005, CallerID(num) is: 2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:3]
Dial(SIP/413-0005, SIP/412) in new stack
[Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412
[Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-0006 is ringing
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, 412,
3) exited non-zero on 'SIP/413-0005'
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Hangup(SIP/413-0005, ) in new stack
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1)
exited non-zero on 'SIP/413-0005'
As you can see on line 3, CallerID(num) was successfully set to 2.
SIP/412 is dialed next. It receives the call, but with 412 as the Caller ID
number - even though the real source of the call was extension 413. The name
I set in CallerID(name) works fine.
My Extensions.conf for that context:
[sipphones]
exten = 412,1,Set(CALLERID(num)=2)
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})
exten = 412,n,Dial(SIP/412)
exten = 412,n,NoOp(${CALLERID(num)})
If I disable sippusers and sippeers in extconfig.conf and put 412 and 413
into sip.conf directly, this code works (ie: the CallerID(num) I set makes it
out to the destination phone properly).
Have you set the fromuser= field in the realtime database?
/O
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Re: [asterisk-users] OT: certificate for softphone
6 nov 2010 kl. 15.30 skrev Hans Witvliet: Hi all, As stated in the subject, slightly off-topic, as it is not directly a Asterisk issue, but more SIP in general Because security in general, and specifically identification becomes more and more a subject for more concern, and Asterisk is capable of doing sip/TLS, i was wondering what more could be done to improve security. Specially softphones, might it be possible to employ etokens or smartcards for holding the certificates needed by TLS? Done before? In the SIP protocol there is support for TLS client certificates, much like in HTTP. Asterisk doesn't support it. You need to put a SIP proxy like Kamailio in front of Asterisk to get this kind of strong authentication. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exceptionally long queue length queuing . . . .
31 okt 2010 kl. 13.43 skrev Paul Belanger: On Sat, Oct 30, 2010 at 6:22 PM, Brian Capouch bri...@palaver.net wrote: I wonder if anyone out there has a perspective on this. There are a welter of tickets out there on the matter, most of them closed. I'm actually able to reproduce this pretty often, for me using IAX2 with IMAP voicemail (google apps) is how. I haven't had much time to debug it, but plan to play more with it the coming weeks. Any update, Paul? /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feature Request for 1.10 - ISDN power-save mode
2 nov 2010 kl. 17.19 skrev Olivier: Hi, In Europe many Telcos implement power-save mode (See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to 'Activation / Deactivation' for more information). Would you agree to have this feature added to the ones already discuused for next Asterisk release ? (See https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010) The projects you see on that list all have resources allocated to them or reasonable close to get allocated by the persons that participated in that meeting - unless you find them in the final categories (3.9 and 3.10). If you have development resources or funding and can create code that works, we are ALWAYS open for contributions, regardless of our lists. Looking forward to your contribution! Best regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk in the third world - Astricon 2010 keynote follow-up
Friends, After listening to Mark Summer's keynote at Astricon (hopefully soon on the Astricon web site) I think we should come back to the discussion he started. Mark talked about using Open Source in general and Asterisk in particular in third world projects as well as in emergencies in other countries. He and Inveneo help groups of people to get a better understanding of how to build network, IP and voice infrastructures. One part is of course learning and managing Asterisk. I do believe many of us wants to help his efforts, but lack the understanding and channels to reach out. I had a very brief discussion with Mark after the keynote and promised to get back to him. My thoughts are that if anyone from these countries try to reach us, we fail to listen and help. Could be language, could be attitude or something else. We can't expect them to have full understanding of net etiquette, the rules of Open Source project management or how to find information themselves (in a language they might not understand fully). The climate in our mailing lists and chat rooms are not always one of understanding, especially if someone copies their english language and attitude from Miami Vice ;-) Do you have any ideas of what could be done from our community? Can we create special forums where we have a different climate, more languages and better understanding? I also think we should copy ISOCs efforts and have a pre-astricon training/workshop for people that Inveneo locate and then invite them to Astricon, funded by grants form community or from somewhere else (since we lack an Asterisk foundation that could help here). I'm sure we can find resources to get them to Astricon and that we can find teachers in the community that are willing to help with this project. I would not hesitate in donating a few days myself. We have enormous powers in our community. If we can gather a small part of that and point it towards these people, we can change the situation for many more, just by doing what we do each day - enjoy building voice solutions and sharing our knowledge. Let's brainstorm for a while! The floor is open. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk distributed device state = res_jabber Versus res_ais
11 aug 2010 kl. 15.49 skrev Leif Madsen: On 10-08-10 04:11 AM, Olle E. Johansson wrote: 26 jul 2010 kl. 18.13 skrev Leif Madsen: On Asterisk 1.6.2, your only option for distributing device state is with res_ais. I've used it in a labbing system and it works well -- the caveat is that your machines need to be on a low latency network (i.e. LAN). With Asterisk 1.8 (currently 1.8.0-beta1) you can use XMPP to distribute your device states over the WAN. I've made it work with the Tigase XMPP server. More information about it can be found in the doc/distributed_devstate-XMPP.txt file. This fall, we're going to implement it using SIP for 1.4 and 1.8/trunk. Look for project pinana. Development will start later this month. Sounds very cool! I look forward to playing around with it. Also thanks for picking a branch name that is not related to fruit or frogs. Thanks for the feedback. I guess the name was a mistake and I'll take it under reconsideration :-) /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk distributed device state = res_jabber Versus res_ais
26 jul 2010 kl. 18.13 skrev Leif Madsen: On 10-07-26 10:45 AM, Mathieu wrote: Hello, as I'm looking for a solution (with asterisk 1.6.2) , my investigations leaded to : - res_ais = libais corosync. (each node need to run corosync / aiexec) - res_jabber = libjabber iksemel. (each node need to be connected on an XMPP server) I've been able to make some successful tests with res_ais on 2 servers but got some CPU issues with corosync after some hours of activity. What's the best solution regarding flexibility and stability and real-time exploitation ? I've got the feeling a good (and old) XMPP server will be more reliable than res_ais which seems to be pretty young. On Asterisk 1.6.2, your only option for distributing device state is with res_ais. I've used it in a labbing system and it works well -- the caveat is that your machines need to be on a low latency network (i.e. LAN). With Asterisk 1.8 (currently 1.8.0-beta1) you can use XMPP to distribute your device states over the WAN. I've made it work with the Tigase XMPP server. More information about it can be found in the doc/distributed_devstate-XMPP.txt file. This fall, we're going to implement it using SIP for 1.4 and 1.8/trunk. Look for project pinana. Development will start later this month. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving CLI Help - was [Re: 1.6.2 No soft hangup?]
Further to Steve Edward's comment, I think things would be more obvious if the help system was improved slightly, for instance: If you were trying to figure out the commands dealing with peers, you would be able to type: *CLI help peer No peer command found. Possible alternatives: iax2 show peer Show details on specific IAX peer iax2 show peers List defined IAX peers sip show peers List defined SIP peers sip show peer Show details on specific SIP peer (and so on, maybe using the [More] option to help it be readable) In this case, if I could use the help system to search on all occurrences of the word hangup in the available commands, I would probably have found it myself instead of bothering the list. THat's a very good idea. Thank you! Now we need someone that codes it :-) /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8
and guru Olle E. Johansson, one that was recognized with a strange smile by all Asterisk developers testing VCC. VCCnet technology includes scalability and security components licensed by Edvina AB in Sweden. Edvina's experience of large scale Unified Communication networks was necessary to build a world-wide network-centric platform for this new service. - We find it exciting to contribute to this new service. Realizing the perfect match between the open IPv6 protocol and the proprietary Dundi technology was an eye-opener. No NAT issues and the possibility to build a worldwide network with service discovery, security and managed QoS will make this a success story. We're proud to contribute to this solution. says Olle E. Johansson, founder of Edvina. The new IAX3 protocol is also really interesting, as it not only combines media and signalling over one port, but now also adds presence, instant messaging, file transfer, printing, database queries, directory services and network management over the same port. It's a one-size-fits-all protocol that will handle all services a user want. The VCCnet network is already in operation, The VCCnet PBX interface will be part of Asterisk 1.8 to be launched later this year and part of the VoxSwitch update Q2 2010. The VCCstore opens June 1st. Development kits are available to Digium authorized VCC development partners today. The VCC technology is patented by Digium and will be operated as a private virtual network on top of the Internet and the ISDN network. For questions and further information, please contact the Digium marketing department at loofli...@digium.com today. A press conference will be held April 1st, 15:00 GSM+1 in VCCconference room 142857 for media representatives. It will be available for one week on vcc://digium.com::conference:142857 for later viewing. VCC, VCCnet, VCCblock, VCCstore, Digium, IAX3, Dundi and Asterisk are trademarks registered by Digium Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtp.conf ports for inbound or outbound?
25 mar 2010 kl. 13.14 skrev Michelle Dupuis: I can't find this in the wiki/email history..but I'm sure it's based asked before. The port range define in rtp.conf - is that for connections initiated by asterisk? Or the port range asterisk listens on? Or both? These are the ports Asterisk use for INCOMING media, the ports we listen on. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] permit/deny in sip.conf iax.conf
24 mar 2010 kl. 16.48 skrev Karl Fife: Steve Edwards wrote: It may not be as intended, but from a user standpoint, it seems logical and convenient to establish policy in [general] and make exceptions in the entities as needed. Right... for when you have one policy. When you have two policies, each that apply to a dozen or more entries in the config file, then it really doesn't help, it harms. Templates solve that problem completely, because each policy can be its own (named!) template, and they can be combined. Since templates are also very easy to use for the single policy case, they are a better solution to teach people (and they're also easier to implement in the configuration code of the module). In other modules created since chan_sip, we've intentionally avoided this problem, and you'll note that in nearly every other module, the [general] section is exactly that; general settings for the module, and not defaults. In my NACL work, I implemented a channel-wide NACL for blacklist purposes. Can you talk more about this? Were your Named ACL's something other than templates? What was/were the specific 'pain point/s' you were trying to assuage? For example did you need something not currently offered in the existing frameworks, for example DNS-resolved hostnames for permitting/restricting registration/connection? Or were you just doing a clever/elaborate/well-implemented setup of the existing frameworks? I for one would love to hear your 10,000 foot concepts and any details you'd be willing to share. Well, I've written several mails and blog entries about this. Many discussions about security in Asterisk has ended with the need for a new concept for ACLs, something that can be manipulated by Asterisk using the C API, by using manager and the CLI. So currently, it's a framework. You can create a named ACL that is used by multiple devices or SIP trunks. In the future, we have the API to build all kind of blacklist/whitelist functions. And I'm open for input on what's needed here. Now we have the framework to build on. http://www.voip-forum.com/asterisk/2010-01/manageable-access-control-lists-asterisk-nacls/ http://svnview.digium.com/svn/asterisk/team/oej/deluxepine-1.4/README.nacl It's something I'm working on just for fun, so it moves slowly forward. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] permit/deny in sip.conf iax.conf
23 mar 2010 kl. 22.20 skrev Kevin P. Fleming: Steve Edwards wrote: It may not be as intended, but from a user standpoint, it seems logical and convenient to establish policy in [general] and make exceptions in the entities as needed. Right... for when you have one policy. When you have two policies, each that apply to a dozen or more entries in the config file, then it really doesn't help, it harms. Templates solve that problem completely, because each policy can be its own (named!) template, and they can be combined. Since templates are also very easy to use for the single policy case, they are a better solution to teach people (and they're also easier to implement in the configuration code of the module). In other modules created since chan_sip, we've intentionally avoided this problem, and you'll note that in nearly every other module, the [general] section is exactly that; general settings for the module, and not defaults. In my NACL work, I implemented a channel-wide NACL for blacklist purposes. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Better SIP security please! Was: (no subject)
21 mar 2010 kl. 18.22 skrev Philipp von Klitzing: Hi Olle! The work I started during Christmas - Named ACL's - is a starting point that other developers can use to develop all kind of schemes. http://www.voip-forum.com/asterisk/2010-01/manageable-access-control-lists -asterisk-nacls/ Very interesting. Doesn't look like this has any chance to secure 1.4 installations though, I am afraid. The code was written both for trunk and 1.4. It won't be included in 1.4 release though, right. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP codec negotiation / manipulation
22 mar 2010 kl. 14.54 skrev Kevin Sandy: On 3/21/2010 4:05 AM, Olle E. Johansson wrote: 17 mar 2010 kl. 16.37 skrev Kevin Sandy: We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second invite requesting G729. However, they proceed to send us a G711 encoded audio stream... They have somewhat acknowledged the problem, but their advice is for us to only accept a single codec in our 200 OK. We don't want to disable either; we have customers using G729, so we'd like to avoid transcoding when possible, but we also do some T38 faxing, which I believe requires G711 to start off. My first thought was to selectively force the codec on inbound calls - if it is for a voice number, use 729, otherwise 711. However, I can't find any way of doing this within Asterisk. (We do have an OpenSIPS server sitting between us and the provider, and I could use OpenSIPS features to do this; however, right now the OpenSIPS server is fairly dumb - it's only proxying traffic between us and the provider and knows nothing about our specific DIDs.) A couple more details in case anyone has seen a similar issue. The provider is Broadvox, and this issue only seems to manifest on calls coming to them via Skype. They claim to not have any direct link with Skype, but it seems odd that the problem would be specific to Skype callers if the call is coming to Broadvox as a standard PSTN call. Is there any way to do this? Am I totally missing something and making a stupid mistake, or making the issue more complicated than it needs to be? The problem here is that you have a proxy in between, so Asterisk can't have separate peer configurations, since all the SIP messages are from the same IP and thus the same peer. I have a branch that implements peer matching in this specific configuration, which means that you can have different codec configurations for different partners even though there's a proxy in front of Asterisk. https://origsvn.digium.com/svn/asterisk/team/oej/pinetree-1.4 Please try this branch and give feedback. There should be some docs in sip.conf for the new matchrule setting. /O I'd be interested in trying this out - but the site doesn't seem to be responding. :) Sorry, gave you the developer URL. Too quick copy and paste... Here's a correct one: http://svn.digium.com/svn/asterisk/team/oej/pinetree-1.4 /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] In Berlin this week? Kamailio/Asterisk community dinner on Thursday
Friends, Daniel and I are running a Kamailio SIP Masterclass this week in Berlin. When travelling around like this, we often invite the community to come and meet us in a nice restaurant. We offer good company and fun discussions about Kamailio, SIP-router.org and Asterisk - but the drinks and food are on you. At least yours :-) Berlin is the city where Sip Express Router was born. Many SER/SIP-router and Kamailio developers live here, so we suspect that you'll find a good set of core developers joining us. Hint: Buying a beer for a developer is generally considered a good thing. Buying too many will affect the commits the next day... The bad code submissions can be reverted easily, so don't worry about it. We'll just have to handle the situation... - Where? The Lemke Brauhaus, Luisenplatz1, 10585 Berlin (close to Schloss Charlottenburg). - Time? 19.00 Berlin time - URL: http://www.brauhaus-lemke.com/index.php?area=4 Please send me a not off-list if you think you can participate, so that we can get a properly sized table. If you want to take a chance, just show up. Either way, you're welcome! This is also a good way to prepare for the VoipAthon - the 24 hour Voip Users Group session. Don't miss that! http://voipathon.org/ The next Asterisk SIP Masterclass will be hosted by Telespeak in the UK. Check their web site for information! I suspect we can find beer or someting compatible in that area too :-) Regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP codec negotiation / manipulation
17 mar 2010 kl. 16.37 skrev Kevin Sandy: We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second invite requesting G729. However, they proceed to send us a G711 encoded audio stream... They have somewhat acknowledged the problem, but their advice is for us to only accept a single codec in our 200 OK. We don't want to disable either; we have customers using G729, so we'd like to avoid transcoding when possible, but we also do some T38 faxing, which I believe requires G711 to start off. My first thought was to selectively force the codec on inbound calls - if it is for a voice number, use 729, otherwise 711. However, I can't find any way of doing this within Asterisk. (We do have an OpenSIPS server sitting between us and the provider, and I could use OpenSIPS features to do this; however, right now the OpenSIPS server is fairly dumb - it's only proxying traffic between us and the provider and knows nothing about our specific DIDs.) A couple more details in case anyone has seen a similar issue. The provider is Broadvox, and this issue only seems to manifest on calls coming to them via Skype. They claim to not have any direct link with Skype, but it seems odd that the problem would be specific to Skype callers if the call is coming to Broadvox as a standard PSTN call. Is there any way to do this? Am I totally missing something and making a stupid mistake, or making the issue more complicated than it needs to be? The problem here is that you have a proxy in between, so Asterisk can't have separate peer configurations, since all the SIP messages are from the same IP and thus the same peer. I have a branch that implements peer matching in this specific configuration, which means that you can have different codec configurations for different partners even though there's a proxy in front of Asterisk. https://origsvn.digium.com/svn/asterisk/team/oej/pinetree-1.4 Please try this branch and give feedback. There should be some docs in sip.conf for the new matchrule setting. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Better SIP security please! Was: (no subject)
19 mar 2010 kl. 03.41 skrev Philipp von Klitzing: Hey hey! My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone quite widely open. Asterisk doesn't differentiate between a hard phone and a soft phone. Although: One could think about enhancing Asterisk security by allowing only a (number of) specific SIP user agent header (vendor, model) for a SIP account - next to a strong password, of course. Or implement something more dynamic like: Read and lock the current (or first) user agent string, and then ping the admin if that changes and request an un- lock/re-auth. Those are interesting ideas. We could implement a timeout for registrations, so that we only accept re-registrations while we have an active registration, and if that expires only accept new registrations after a timeout. This will delay access at reboots of the Asterisk server though. Does Asterisk 1.6 have anything in it that can automatically block out an attacking IP, say if it receives several 20 or so failed attempts from that IP in x minutes? It would still be important to have a sip.conf paramter in 1.4 that is similar to delayreject in iax.conf! One of my system has been scanned 3 times in the past days, and it takes just a little over a minute for a 10.000 account registration scan. The work I started during Christmas - Named ACL's - is a starting point that other developers can use to develop all kind of schemes. http://www.voip-forum.com/asterisk/2010-01/manageable-access-control-lists-asterisk-nacls/ /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf payload 100
12 mar 2010 kl. 10.45 skrev Katerina Borin: Probably has anyone idea how dtmf payload type could be changed in Asterisk say to 100? On Wed, Mar 10, 2010 at 2:53 PM, Katerina Borin katerin.bo...@gmail.com wrote: Hello, I encountered the dtmf problem between my asterisk box (1.4.23) and suppliers gateway (unknown vendor). I have dtmf mode set to rfc2833 and it alway worked till supplier has changed something. Now I receive from him dtmf payload 100. With the second supplier which sends dtmf with payload type 101 everything works. in cli I get this message as dtmf is entered rtp.c:1287 ast_rtp_read: Unknown RTP codec 100 received from 'suppliers IP' Is there any way to get asterisk understand dtmf payload type 100? If they have declared it correctly in the SDP, we will understand. Since Asterisk doesn't recognize the codec, I belive they have a bug in their system. In order for us to find out if Asterisk is doing wrong or if we can blame their system, we need to see the INVITE or 200 OK from their end. The information you have provided here is not enough. THanks, /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Trunk with multiple remote ip-addresses
12 mar 2010 kl. 12.01 skrev Klaus Darilion: Am 02.03.2010 13:29, schrieb Magnus Benngård: Hi! Did a setup of 2 peers as Klaus suggested, it worked thx! Has anyone thought about the possibility to add multiple ip/hosts to host=? I my case: host=130.244.190.42,130.244.190.46 or host=sip-corporate1.tele2.se,sip-corporate2.tele2.se Step 1 could be to send to the first ip/host and accept from both. Step 2 could be round-robin send if both are up and alive... IMO this would be a nice feature. Check my peerfailover branch. Btw, did try trunk version, no support for multiple SRV records there. IIRC correctly there is a patch on the bugtracker for SRV handling, but I do not know if that patch would fix this too. I haven't seen that. Interesting. /O regards klaus Am 02.03.2010 08:50, schrieb Magnus Benngård: Hi, Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No problem to get outgoing calls to work but i have some problems with incoming. Did set srvlookup=yes in sip.conf. Sending all outgoing calls to sip-corporate.tele2.se which is either sip-corporate1.tele2.se (130.244.190.42) or sip-corporate1.tele2.se (130.244.190.46). If i do a sip show peer Tele2, I see that Asterisk has chosen one of them: ToHost : sip-corporate.tele2.se Addr-IP : 130.244.190.46 Port 5060 Now my problems starts, when Tele2 sends a call to my Asterisk, the call can come frome any of those two ip-adresses. If it comes from 130.244.190.46 everything if fine, but if it comes from 130.244.190.42: [Mar 2 08:46:03] NOTICE[1372]: chan_sip.c:19167 handle_request_invite: Failed to authenticate! I thought srvlookup=yes should take care about that, but then i read a little bit more and found: Note: Asterisk only uses the first host in SRV records. :( Hi Magnus! Asterisk does not support multiple SRV records (expcet there were some recent changes which I missed) - it takes one of the most priors and use it all the time. Thus, in your scenario you have to specify the possible inbound sources manually as peers: [tele2-1] type=peer host=130.244.190.42 context=fromTele2 ... [tele2-2] type=peer host=130.244.190.46 context=fromTele2 ... regards klaus Can anyone plz give me some hint howto solve my problem? Regards, Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way for a peer to clear its registration from a server?
11 mar 2010 kl. 15.17 skrev Philipp von Klitzing: Is there a way for a client to tell a server where it is registered to remove the registration? Yes, it needs to send an UNREGISTER sip message. There's actually not an UNREGISTER method in SIP. As Kevin stated, you send a REGISTER with a zero expiry to cancel a current registration. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Management API
8 mar 2010 kl. 11.13 skrev Peter Childs: On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote: At an Asterisk CLI use the command manager show commands. Life is rarely that simple, and this does not really answer the question. Oh and Channel can mean different things in different contexts ie Channel in a PlayDTMF command means a Call to play the DTMF on, where as Channel in a Originate command means the Device to place the call on so you can't use the same input for both commands (or can you?) I agree that it's kind of stupid. I cleared up some of that mess in 1.6.x, but not all. And the changes hurted a lot of existing applications, so I'm careful not to mess around too much with AMI again. The most important part is that we don't allow reuse of existing headers for new things in new actions and events. I've been trying to watch over manager in order to disallow misuse, but development is fast and it's easy to miss a commit or a review... /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPit 26 in Sweden - organized by Edvina
Friends, SIPit is the main interoperability event for all things SIP. It's organized by the SIP Forum and creates good feedback to the IETF. Asterisk has been participating in SIPit during many years and in many variants - videocaps, Marc Blanchet's IPv6 branch and the standard Digium releases. All these tests has lead to a large amount of improvements for Asterisk and have helped us to build a network with other developers in the business, a network which helps when we have bugs that involve interoperability with these devices or servers. SIPit is important for Asterisk, and thus it is important for everyone in the Asterisk community. Now, when we are working on the next long-term release (1.8) we really need to test and make sure that we interoperate. New stuff, like Terry's SRTP branch and the call completion and caller ID update work needs testing. * Interoperability drives the TCP/IP business What drives the TCP/IP business is simply stated in one word - INTEROPERABILITY. Without open network standards, the VoIP business would not be as large as it is today. Without working and tested standards, it would not work at all. Asterisk, as an Open Source platform is in the middle of this business. We simply have to interoperate with all kinds of phones, servers and services out there. * SIPit 26 - organized by Edvina SIPit is organized by the SIP Forum and every SIPit - two per year - is hosted by a company. During good times, the large vendors has taken care of this. In the current climate, it's hard to get the needed resources - time and money - from these vendors. SIPit 25, yes the 25th in a successful series, was organized by the Interoperability Labs at University of New Hampshire in September 2009. Digium helped out by providing the PBX infrastructure for the event. The next one, SIPit 26, will be in Stockholm, Sweden - hosted by myself with support from Tandberg and sponsored by a set of companies. * SIP interoperability is a requirement - for you, your customers, your business So why I am doing this? The business needs it, Asterisk needs it and I need it. Without SIPit, you will not get good products that work together. Without SIPit, tests will be limited to certification of a very limited set of functions by different vendors. I believe that this will lead to unhealthy market domination in the implementations, something that does not benefit the customers. And besides, after creating Astricon in the US and a series of other conferences in Sweden, I have experience of organizing events. I am crazy enough to step forward and take the risk, since I really strongly believe that we all benefit from this. Now I need your support too. * Additions to your already filled-up TODO-list: - If you develop SIP software, make sure you register and attend. - If you buy SIP devices or software, ask the vendors if they attend SIPit and encourage them to participate - If you have collegues that work with SIP development, please forward this mail to them! Short facts: - Date: May 17-21 2010 (very beautiful season in Sweden!) - Location: Kista, Stockholm, Sweden - Host: Edvina and Tandberg - Sponsors: Ingate, Intertex, .se, Telio, Snom We are currently working to set the price and to be able open for registration. SIPit 26 has a Facebook event page at http://www.facebook.com/event.php?eid=340634688354 and a Twitter stream: http://twitter.com/sipit26 where you will get updates and be able to find links to the host web site when it opens. The main web site for SIPit is http://www.sipit.net - a site that explains what will happen during this week and why you should attend as a developer. If you have any questions or suggestions, please don't hesitate to contact me. Thanks for your support! Regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup
27 feb 2010 kl. 08.26 skrev Olle E. Johansson: 26 feb 2010 kl. 22.02 skrev JT: Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat of a band-aid to the issue. But in my observations there is one clear indicator that I am shocked is not used. When I have done this test - pulling the network cable on a device during a call - Asterisk actually reports that the SIP device has become unreachable within seconds of the device's removal. Now one would think, just like a regular phone company, if one device became unresponsive (unreachable), the call would be automatically dropped. Like unplugging from a POTS while on a call. So why would Asterisk not use the following logic: Is Device reachable? Yes - Do nothing No - Close all calls bridged to device Seems that would solve the issue quickly and cleanly... perhaps with the RTP timeout being an additional measure of safety Is this an issue present in the latest version of Asterisk? My hope was it was simply an older bug, fixed at some later trunk. If there's a reason to send SIP messages during the call and they fail, the call WILL be hung up. Reading the 1.4 RTP source code, I don't think we're checking the return codes of the network writes. Now, that can be very tricky. For a call with NAT, we will have to send packets that fail until we receive something from the other end. I am just brainstorming here, but we could have a flag set when we've received RTp packets from the other end and from that moment start reacting on the result codes of the sendto() call. If it's indicating network issues, we could possibly have an option to tear the call down after a certain amount of failures. And no, I can't explain why someone hasn't thought of that. I think it would be a good addition. And after a few hours of hacking I know more. If the incoming channel dies, there will be no attempts at sending, so we won't have any network issues at all. The RTP channel in Asterisk is clocked on incoming media. The RTP timeouts we have today is the only solution for normally bridged calls. The p2p rtp bridge behaves a bit differently and I think I found a bug in it, so I will have to investigate that part a bit more. Now, we could hang up calls based on device status if needed. I have part of that code in the peerfailover branch. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup
26 feb 2010 kl. 22.02 skrev JT: Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat of a band-aid to the issue. But in my observations there is one clear indicator that I am shocked is not used. When I have done this test - pulling the network cable on a device during a call - Asterisk actually reports that the SIP device has become unreachable within seconds of the device's removal. Now one would think, just like a regular phone company, if one device became unresponsive (unreachable), the call would be automatically dropped. Like unplugging from a POTS while on a call. So why would Asterisk not use the following logic: Is Device reachable? Yes - Do nothing No - Close all calls bridged to device Seems that would solve the issue quickly and cleanly... perhaps with the RTP timeout being an additional measure of safety Is this an issue present in the latest version of Asterisk? My hope was it was simply an older bug, fixed at some later trunk. If there's a reason to send SIP messages during the call and they fail, the call WILL be hung up. Reading the 1.4 RTP source code, I don't think we're checking the return codes of the network writes. Now, that can be very tricky. For a call with NAT, we will have to send packets that fail until we receive something from the other end. I am just brainstorming here, but we could have a flag set when we've received RTp packets from the other end and from that moment start reacting on the result codes of the sendto() call. If it's indicating network issues, we could possibly have an option to tear the call down after a certain amount of failures. And no, I can't explain why someone hasn't thought of that. I think it would be a good addition. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] directrtp with SIP + H.323
24 feb 2010 kl. 01.22 skrev Kristian Kielhofner: On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis supp...@ocg.ca wrote: We're creating a SIP gateway for a client that will take one leg of a call in via SIP, and out the other side via H.323. To minimize load on the gateway, we would like to have the RTP stream bypass the gatewayy altogether (directrtp/reinvite). Is this possible with these to protocols? Thanks Yate claims it can do this: http://yate.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy There are two ways - either by reinvites, which according to Kevin won't work with H323, or by doing it right in the call setup. If we did that, we would stumble into the same problem as we have with this function in SIP - which goes all back to the media negotiation framework (see http://www.voip-forum.com/asterisk/2010-02/asterisk-18-lts-wishlist-1-media-negiotiation-framework/ ). Asterisk currently just communicates an answered call as answered over the bridge without any attributes. This is the reason why the code has been marked experimental for many releases and no one has solved it. In order for this to work, you either need exactly the same codec attributes or a way to handle the ANSWER control frame (like John Martin did in the videocaps branch). The hooks are all there if you want to experiment with this in the H.323 channel. It's certainly possible. But it is not a function I would support generally (which is why the directrtp call setup function remains experimental). /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
23 feb 2010 kl. 20.18 skrev Matt Riddell: The responses from the Asterisk manager on your machine start providing responses of no account code when calls are initiated at a higher rate. Where's the bug report id? I haven't heard about this limit. I don't know what it is, but we should at least be able to accept the originate requests in asynch mode, put them on a queue and process them in a separate thread (which can be configurable in manager.conf). This is just brainstorming - but first, let's try to find out if the limit you believe in exists in the code or is just the effect of something else. /O On 24/02/2010, at 12:59 AM, CDR vene...@gmail.com wrote: My dear friend Matt Riddell insists that the Manager only can dial 5 calls per seconds, which I find ridiculous. Is there a way to prove him wrong and have him lift the limit that has been plaguing the life of us users of SineDialer and SmoothTorrque Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio to remote AGI server
22 feb 2010 kl. 07.23 skrev Tilghman Lesher:
open audio {tcp|udp} hostname portno
close audio
If you design something now, I would strongly suggest that we stop using
audio as an attribute. Each call will have multiple media streams - and
already have. You need to be able to select which one, and possibly open
multiple streams - audio, video, fax, text. In the future, we'll hopefully have
the ability to run multiple of each category, so I would not design this
feature for a single audio stream to be open for future use.
Just my 10 öre. :-)
/O
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Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup
23 feb 2010 kl. 03.18 skrev Kevin P. Fleming: Kirill 'Big K' Katsnelson wrote: The caveat here is that it is perfectly normal NOT to transmit any RTP data in case of long silence. This is why the SIP timers were introduced in the first place: there is no correct way to detect when the client is going away, as no activity is a good session state. That's only true when Asterisk tells the other endpoint that it is allowed to use voice activity detection and silence suppression, which at this point it does not do. In spite of that, there are many endpoints that do it anyway, which then causes strange problems on calls, including calls getting dropped if an RTP timeout is in use. Well, the headers we use are note really standardized, at least I could not find them. In the RTP rfc's it's perfectly legal to just have gaps in the timestamps and stop sending. However, as both me and Kevin stated, Asterisk does not support it. On most phones, you can disable silence suppression in the configuration. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup
23 feb 2010 kl. 01.47 skrev Kirill 'Big K' Katsnelson: On 100222 1313, JT wrote: When a SIP device dials another SIP device...Asterisk connects the calls and displays the channel information. If one of those SIP devices hangs up, Asterisk receives the hangup notice and disconnects the call/channel. However - what does Asterisk do when the network cable is unplugged from one of the SIP devices...?! Jared already mentioned SIP session timers, which are supported starting with 1.6. Here's my experience. While I am running 1.6, the software stack that is used for agent softphone (PJSIP) does not support the session timers. If the softphone crashes in a call, the call would get stuck exactly as you describe. I am working around this problem by setting rtp timeouts in sip.conf: [general] rtptimeout=10 rtpholdtimeout=300 This means that if RTP flow stops while the agent is in the call, the call will be disconnected in 10 seconds. If the call was put on hold by the agent, it will be disconnected in 300 seconds. Your timeouts may vary. The caveat here is that it is perfectly normal NOT to transmit any RTP data in case of long silence. Not in Asterisk - we do not really support silence suppression. The recommendation is to turn it off on the phones. This is why the SIP timers were introduced in the first place: there is no correct way to detect when the client is going away, as no activity is a good session state. I am able to get away with the small timeout because I set the PJSIP client to always transmit RTP, by turning off voice activity detection feature (VAD). If you want to support that feature, set rtptimeout as high as for how long you allow absolute silence on the line without disconnecting it. Just to complete this discussion - we also have the absolute timeout that is a lifesaver in many cases. If you set this to a time that's larger than the normal calls, Asterisk will hang up the call. I very often set it to two hours, just to make sure that if anything strange happens, all calls will be cancelled out at some point. /O -- _ -- 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
