[asterisk-users] IAX2 trunk on IPV6
Hi, I have installed asterisk-1.8.25.0 on an Ubuntu server which has both an ipv6 ip and ipv4 ip(real ip) assigned. And I have a client ubuntu with only ipv4 ip(local ip) installed asterisk-1.8.25.0 . I want to configure the client asterisk with the server asterisk as IAX2 peer and want to connect to the IPV6 ip. I bind the server with ipv6 and also sending the registration request from the client(peer) to the ipv6 address. But its not peering. following is the client's iax.conf register = peer1:peer1pass@[IPV6]:port [peer1] type=peer context=topeer username=peer1 secret=peer1pass trunk=yes host=XXX.XXX.XXX.XXX port= disallow=all allow=g729:40,g723:30 qualify=yes Also my confusion is what value will be in 'host' property. I assigned as host=[IPV6]...but it shows error. Can anyone help with this issue. Thanks in advance -- _ -- 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's internal database
i would like to read information from a file (txt) On Mon, Apr 28, 2014 at 9:29 PM, Rusty Newton rnew...@digium.com wrote: On Thu, Apr 24, 2014 at 6:34 AM, binary dreamer dreamer.bin...@gmail.com wrote: hello everyone. I am running plain asterisk and I am using asterisk's internal database for: -phonebook -blacklist numbers instead of having to update the database of new entry or delete an entry, is it possible to have it in an external file such as txt? so every new entry/deletion will take place there. Are you wanting to swap out Asterisk's internal database with a different data storage interface? If so, that isn't possible as far as I know. If you are wanting to just read information from a file into Asterisk variables.. there may be other ways to do what you want. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://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
[asterisk-users] Destruction of SIP dialog for OPTIONS requests
Hi all, I'd like to verify whether an Asterisk behaviour is expected or not, and ask for advice for the best solution. I have Asterisk 1.8.17.0 on debian wheezy, listening on UDP and TCP 5060, and TLS 5061. Asterisk is part of a dispatcher set in Kamailio (4.1.3), and is marked as AP (Active Probing): this means that Kamailio sends an OPTIONS request every N seconds to verify Asterisk is available. When I try to use TCP or TLS, after some time, Asterisk is marked as Inactive. For what I've seen, the reason is that after 32 after the first OPTIONS, Asterisk destroys the related dialog, and stops replying to the OPTIONS requests on the existing TCP socket. Kamailio times out, and re-opens a new TCP socket when the next probe is due. This works again, and probing is successful, until Asterisk destroys the dialog again. I've configured Asterisk so that is replies with a 200 OK. Is there a way I can avoid the dialog destruction, and let Kamailio use the same TCP socket for whatever time is necessary, rather than about 30? Is my Asterisk behaving as expected? What's your advice for this type of configuration? Thanks in advance, -- Giacomo Vacca -- _ -- 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] IAX2 trunk on IPV6
On Tue, Apr 29, 2014 at 1:06 AM, Xengis Khan xengisk...@gmail.com wrote: Hi, I have installed asterisk-1.8.25.0 on an Ubuntu server which has both an ipv6 ip and ipv4 ip(real ip) assigned. And I have a client ubuntu with only ipv4 ip(local ip) installed asterisk-1.8.25.0 . I want to configure the client asterisk with the server asterisk as IAX2 peer and want to connect to the IPV6 ip. I bind the server with ipv6 and also sending the registration request from the client(peer) to the ipv6 address. But its not peering. following is the client's iax.conf register = peer1:peer1pass@[IPV6]:port [peer1] type=peer context=topeer username=peer1 secret=peer1pass trunk=yes host=XXX.XXX.XXX.XXX port= disallow=all allow=g729:40,g723:30 qualify=yes Also my confusion is what value will be in 'host' property. I assigned as host=[IPV6]...but it shows error. Can anyone help with this issue. IAX2 does not support IPv6 in that version of Asterisk. IPv6 support was added to chan_iax2 in Asterisk 12 [1]. [1] https://wiki.asterisk.org/wiki/display/AST/New+in+12 -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
[asterisk-users] SIP subscribe with multi-server registration
This may be a more phone-specific question, but figured I'd ask to see if someone has experience with this. I have a SIP phone (Polycom) configured with two lines registered to two different Asterisk servers. I have successfully configured SIP subscriptions to watch a different phone registered to server 1. if I try to watch a phone for server 2, the SIP subscription is going to server 1 (and failing). Is there a way to properly configure this so the subscriptions go to the correct place? The only thing I can think of to solve it is to put a proxy in between and route the subscription requests to each server as needed, and I'm not even sure if that is possible (and would require me to learn a LOT more about opensips). Thanks, Josh -- _ -- 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] RFC 4662 in asterisk 10.12.1
Hello, Is there an implementation for the RFC 4662 for asterisk 10? I found a patch for asterisk 1.8 but nothing for asterisk 10.12. The RFC: This document presents an extension to the Session Initiation Protocol (SIP)-Specific Event Notification mechanism for subscribing to a homogeneous list of resources. Instead of sending a SUBSCRIBE for each resource individually, the subscriber can subscribe to an entire list and then receive notifications when the state of any of the resources in the list changes. I have a Panasonic KX-UT133 and this phone use this method to BLF. Anyone worked with this issue? 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
[asterisk-users] SQlite3 realtime
I just finished migrating our web interface from Mysql to SQlite3 and everything seems to be working fine. I just have one detail. The following keeps appearing on my logs: [Apr 29 13:09:32] WARNING[30494]: res_config_sqlite3.c:520 realtime_sqlite3_execute_handle: Could not execute 'UPDATE sip_buddies SET ipaddr = '192.168.0.52', port = '5060', regseconds = '1398795032', defaultuser = '112', useragent = 'Zoiper r21999', lastms WHERE name = '112'': near 112: syntax error5060;rinst [Apr 29 13:10:26] WARNING[30494]: res_config_sqlite3.c:520 realtime_sqlite3_execute_handle: Could not execute 'UPDATE sip_buddies SET ipaddr = '192.168.0.52', port = '5060', regseconds = '1398795086', defaultuser = '112', useragent = 'Zoiper r21999', lastms WHERE name = '112'': near 112: syntax error5060;rinst I guess these are the realtime updates when an extension registers with Asterisk. I really do not know why it is trying to update the defaultuser field but the problem seems to be that the SQL code for updating the table is has a syntax error. So far everything is working, just keep getting this messages on the log file. Should I report this as a bug? We are using Asterisk 11.7.0 on CentOS 6 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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 support for h.324m
Hello, If anyone has successfully compiled asterisk with: app_rtsp codec_amr mp4_play mp4_save app_transcode h324m_call Please share the versions of OS software, and libraries used. Lets make this thread useful so that all tried and tested video resources of asterisk can be found in one place for ease of access and later reference. 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
[asterisk-users] CBAnn channel not going away in Asterisk 12
After an upgrade to Asterisk 12, I'm collecting channels. When I enter and then exit a conference room, I see: -- CBAnn/207-067f;1 Playing 'confbridge-leave.slin' (language 'en') -- Channel CBAnn/207-067f;2 joined 'softmix' base-bridge 5edb1920-3774-4ba3-8c4d-23e8fd04519c -- Channel CBAnn/207-067f;2 left 'softmix' base-bridge 5edb1920-3774-4ba3-8c4d-23e8fd04519c I'd expect those channel to immediately go away, but they just stay around: asterisk*CLI core show channel CBAnn/207-067f;1 -- General -- Name: CBAnn/207-067f;1 Type: CBAnn UniqueID: 1398809161.20186 LinkedID: 1398809161.20186 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) NativeFormats: (nothing) WriteFormat: unknown ReadFormat: unknown WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 0h1m3s Bridge ID: (Not bridged) -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (Empty) Call Identifer: (None) Variables: [Apr 29 18:07:04] ERROR[21102]: cdr.c:3106 ast_cdr_serialize_variables: Unable to find CDR for channel CBAnn/207-067f;1 asterisk*CLI core show channel CBAnn/207-067f;2 -- General -- Name: CBAnn/207-067f;2 Type: CBAnn UniqueID: 1398809161.20187 LinkedID: 1398809161.20186 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 0h3m30s Bridge ID: (Not bridged) -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (Empty) Call Identifer: (None) Variables: [Apr 29 18:09:31] ERROR[21102]: cdr.c:3106 ast_cdr_serialize_variables: Unable to find CDR for channel CBAnn/207-067f;2 -- _ -- 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] CBAnn channel not going away in Asterisk 12
On Tue, Apr 29, 2014 at 5:10 PM, Richard Kenner ken...@gnat.com wrote: After an upgrade to Asterisk 12, I'm collecting channels. When I enter and then exit a conference room, I see: -- CBAnn/207-067f;1 Playing 'confbridge-leave.slin' (language 'en') -- Channel CBAnn/207-067f;2 joined 'softmix' base-bridge 5edb1920-3774-4ba3-8c4d-23e8fd04519c -- Channel CBAnn/207-067f;2 left 'softmix' base-bridge 5edb1920-3774-4ba3-8c4d-23e8fd04519c I'd expect those channel to immediately go away, but they just stay around: asterisk*CLI core show channel CBAnn/207-067f;1 -- General -- Name: CBAnn/207-067f;1 Type: CBAnn UniqueID: 1398809161.20186 LinkedID: 1398809161.20186 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) NativeFormats: (nothing) WriteFormat: unknown ReadFormat: unknown WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 0h1m3s Bridge ID: (Not bridged) -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (Empty) Call Identifer: (None) Variables: [Apr 29 18:07:04] ERROR[21102]: cdr.c:3106 ast_cdr_serialize_variables: Unable to find CDR for channel CBAnn/207-067f;1 asterisk*CLI core show channel CBAnn/207-067f;2 -- General -- Name: CBAnn/207-067f;2 Type: CBAnn UniqueID: 1398809161.20187 LinkedID: 1398809161.20186 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 0h3m30s Bridge ID: (Not bridged) -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (Empty) Call Identifer: (None) Variables: [Apr 29 18:09:31] ERROR[21102]: cdr.c:3106 ast_cdr_serialize_variables: Unable to find CDR for channel CBAnn/207-067f;2 The announcer channel is not supposed to go away while the conference exists so it can be reused for the next sound to play into the conference. The announcer channel joins/leaves the conference as it has sounds to play. If the channel still hangs around after the conference is destroyed then there is a problem. Richard -- _ -- 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 support for h.324m
On 29-04-14 20:41, [Digital^Dude] ® wrote: Hello, If anyone has successfully compiled asterisk with: app_rtsp codec_amr mp4_play mp4_save app_transcode h324m_call Please share the versions of OS software, and libraries used. Lets make this thread useful so that all tried and tested video resources of asterisk can be found in one place for ease of access and later reference. I haven't but the guys you could talk to are the (former Fontventa?) folks at http://www.medooze.com/products/h324m-stack.aspx Source code/binaries can be viewed/downloaded at: http://sourceforge.net/p/asteriskvideo/ http://sourceforge.net/projects/mcumediaserver/ The old Fontventa AMR patch does not apply to Asterisk 11. I tried to port it to Asterisk 11 but couldn't get a call going. If you are a developer focused on Asterisk 11 and want to have a look at the AMR patch let me know and I'll email it to you. HTH, Patrick -- _ -- 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] CBAnn channel not going away in Asterisk 12
The announcer channel joins/leaves the conference as it has sounds to play. If the channel still hangs around after the conference is destroyed then there is a problem. There's a problem. ;-) But thanks for pointing to how that's supposed to be handled. -- _ -- 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] CBAnn channel not going away in Asterisk 12
If the channel still hangs around after the conference is destroyed then there is a problem. Am I missing something obvious: I'm looking in the confbridge_exec function. I see a conference = NULL line, but no attempt to free that structure, which is what I understand will destroy the playback channel. So where it is freed? -- _ -- 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] Inbound DAHDI Error
Hello, I am trying to diagnose an intermittent error when a call comes in over our PRI lines. The problem appears random, however I have feeling it has something to do with the call volume, as the frequency increases with more calls on the system. I am not an expert when it comes to reading the PRI Span Debug statements but here is a call that had a problem and I bolded, italicized, and underlined the part of the debug statement that looks odd (listed under PRI Debug Output (failed call)). Any help is appreciated. Thanks, Bryce *Version(s):* Asterisk 11.8.1, installed from the Digium YUM Repositories DAHDI Version: 2.9.0 Digium Card: Wildcard TE235 (VPMOCT064) OS: CentOS 6.5 *My Observations:* When I have the problem, the only way I see that Asterisk received a signal on my PRI lines was through the pri debug statements, I don’t see anything being hit in the dialplan (for instance the NoOp at the start of my sub-dial-cudatel-extension sub context). Is there another tool I should be using to debug this issue? *PRI Debug Output (failed call):* PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=73 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 23832/0x5D18) (Sent from originator) PRI Span: 1 Message Type: SETUP (5) PRI Span: 1 [04 03 80 90 a2] PRI Span: 1 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 1 User information layer 1: u-Law (34) PRI Span: 1 [18 03 a1 83 81] PRI Span: 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0 PRI Span: 1ChanSel: As indicated in following octets PRI Span: 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 PRI Span: 1Ext: 1 Channel: 1 Type: CPE] PRI Span: 1 [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 4f 4d 41 58 20 43 4f 52 50 20 4e 20 47 53 4d] PRI Span: 1 Facility (len=31, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0F, 'source_caller_name' ] PRI Span: 1 [6c 0c 21 83 32 35 33 33 38 30 35 35 39 31] PRI Span: 1 Calling Party Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) PRI Span: 1 Presentation: Presentation allowed, Network provided (3) 'calling_caller_id' ] PRI Span: 1 [70 0b a1 32 35 33 38 37 32 32 33 30 30] PRI Span: 1 Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'dest_number' ] PRI Span: 1 -- Making new call for cref 23832 PRI Span: 1 Received message for call 0x7f7a900012f0 on link 0x1a3cf70 TEI/SAPI 0/0 PRI Span: 1 -- Processing Q.931 Call Setup PRI Span: 1 -- Processing IE 4 (cs0, Bearer Capability) PRI Span: 1 -- Processing IE 24 (cs0, Channel ID) PRI Span: 1 -- Processing IE 28 (cs0, Facility) PRI Span: 1 -- Processing IE 108 (cs0, Calling Party Number) PRI Span: 1 -- Processing IE 112 (cs0, Called Party Number) PRI Span: 1 -- Delayed processing IE 28 (cs0, Facility) PRI Span: 1 ASN.1 dump PRI Span: 1 Context Specific [11 0x0B] 8B Len:1 01 PRI Span: 1 00 - ~ PRI Span: 1 Context Specific/C [1 0x01] A1 Len:23 17 PRI Span: 1 Integer(2 0x02) 02 Len:1 01 PRI Span: 1 01 - ~ PRI Span: 1 Integer(2 0x02) 02 Len:1 01 PRI Span: 1 00 - ~ PRI Span: 1 Context Specific [0 0x00] 80 Len:15 0F PRI Span: 1 4F 4D 41 58 20 43 4F 52-50 20 4E 20 47 53 4D - source_caller_name PRI Span: 1 ASN.1 end PRI Span: 1 interpretation Context Specific [11 0x0B] = 0 0x PRI Span: 1 INVOKE Component Context Specific/C [1 0x01] PRI Span: 1 invokeId Integer(2 0x02) = 1 0x0001 PRI Span: 1 operationValue Integer(2 0x02) = 0 0x PRI Span: 1 operationValue = ROSE_QSIG_CallingName PRI Span: 1 callingName Name PRI Span: 1 namePresentationAllowedSimple Context Specific [0 0x00] = PRI Span: 1 4F 4D 41 58 20 43 4F 52-50 20 4E 20 47 53 4D - source_caller_name PRI Span: 1 q931.c:8646 post_handle_q931_message: Call 23832 enters state 6 (Call Present). Hold state: Idle Span 1: Processing event PRI_EVENT_RING(5) *PRI Span: 1 q931.c:7135 q931_hangup: Hangup other cref:23832* *PRI Span: 1 q931.c:6892 __q931_hangup: ourstate Call Present, peerstate Call Initiated, hold-state Idle* *PRI Span: 1 q931.c:6081 q931_disconnect: Call 23832 enters state 11 (Disconnect Request). Hold state: Idle* PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=73 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 23832/0x5D18) (Sent from originator) PRI Span: 1 Message Type: SETUP (5) PRI Span: 1 [04 03 80 90 a2] PRI Span: 1 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 1
