[asterisk-users] pyAsterisk: how to gracefully exit from event loop
Hello, I'm using py-Asterisk 0.5.3. I'm trying to use it along a Tkinter-based GUI so I've dedicated a thread for reading incoming AMI events. Which is the preferred way to gracefully exit from an event loop ? More precisely: This thread is waiting for input events with Manager._read_packet() method. Doc (see [1]) says: ... infinite loop running BaseManager.read() until an exception occurs (for example, SystemExit is raised) or until sys.exit() is called The best method I could find is to close Manager connection and catch exception. Any advice ? Regards [1] http://py-asterisk.googlecode.com/hg-history/242456f432f2fa2727b26d648bcf7dc502fdcc51/doc/GUIDE.html -- _ -- 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] Realtime Call Queues : call members in certain order
Hello, I'm using MySQL realtime Call Queues (table /queues/ and table /queue_members/). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy /lineair /and I put the members into the table /queue_members/ in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout | monitor_type | monitor_format | queue_youarenext | queue_thereare | queue_callswaiting | queue_holdtime | queue_minutes | queue_seconds | queue_lessthan | queue_thankyou | queue_reporthold | announce_frequency | announce_round_seconds | announce_holdtime | announce_position | retry | wrapuptime | maxlen | servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus | eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart | periodic_announce | periodic_announce_frequency | ringinuse | +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ | queue6 | default| NULL | | 12 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | 30 | NULL | No | yes | 5 | 10 | 0 | NULL | linear | strict| strict | NULL | NULL| NULL |NULL | NULL | no | | 0 | no| +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ and queue members : +--++++-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++++-++ | 44 | queuemem4 | queue6 | SIP/queuemem4 | 0 | NULL | | 45 | queuemem2 | queue6 | SIP/queuemem2 | 0 | NULL | | 46 | queuemem5 | queue6 | SIP/queuemem5 | 0 | NULL | | 47 | queuemem1 | queue6 | SIP/queuemem1 | 0 | NULL | | 48 | queuemem10 | queue6 | SIP/queuemem10 | 0 | NULL | | 49 | queuemem18 | queue6 | SIP/queuemem18 | 0 | NULL | | 50 | queuemem17 | queue6 | SIP/queuemem17 | 0 | NULL | | 51 | queuemem12 | queue6 | SIP/queuemem12 | 0 | NULL | | 52 | queuemem16 | queue6 | SIP/queuemem16 | 0 | NULL | | 53 | queuemem13 | queue6 | SIP/queuemem13 | 0 | NULL | +--++++-++ You can see that the member /queuemem4/ is first in line to be rang (has the first and lowest uniqueid in the table). But the first member that is being rang, is /queuemem//1/. How come ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Support of callto or tel protocols in MS Office ?
Hello, Has someone successfully configured support of either callto or tel protocol in MS Office in general or MS Office Online's Outlook specifically ? (I'm referring here in Outlook client embedded in MS Office cloud service). If positive, what are the basic steps to enable such feature (clicking on a contact phone number triggers whatever program is attached to tel/callto protocol in Windows Registry) ? Regards -- _ -- 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] g726 transcoding
On 11/02/14 18:45, Dave Platt wrote: Just checking the transcoding on our Asterisk boxes and I get the following results. I have the g726, ilbc and lpc10 formats and codecs enabled in 'make menuselect' so I dont understand why its showing as no translation path. Any ideas? Are the modules actually loaded? Try doing a module show and see if the codec modules actually show up as having been loaded. If not check your modules.conf file and see if they've been disabled, and check your Asterisk modules directory to confirm that they were actually installed. Yes that was the problem. I posted an update yesterday but it got sent from the wrong account so didnt make it to the list. When I installed the system I had disabled a lot of the modules from load which were not used and the codecs were part of that. -- _ -- 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 Call Queues : call members in certain order
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, February 12, 2014 3:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Realtime Call Queues : call members in certain order Hello, I'm using MySQL realtime Call Queues (table queues and table queue_members). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy lineair and I put the members into the table queue_members in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout | monitor_type | monitor_format | queue_youarenext | queue_thereare | queue_callswaiting | queue_holdtime | queue_minutes | queue_seconds | queue_lessthan | queue_thankyou | queue_reporthold | announce_frequency | announce_round_seconds | announce_holdtime | announce_position | retry | wrapuptime | maxlen | servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus | eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart | periodic_announce | periodic_announce_frequency | ringinuse | +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ | queue6 | default| NULL | | 12 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | 30 | NULL | No| yes | 5 | 10 | 0 | NULL | linear | strict | strict | NULL | NULL| NULL | NULL | NULL | no | | 0 | no| +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ and queue members : +--++++-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++++-++ | 44 | queuemem4 | queue6 | SIP/queuemem4 | 0 | NULL | | 45 | queuemem2 | queue6 | SIP/queuemem2 | 0 | NULL | | 46 | queuemem5 | queue6 | SIP/queuemem5 | 0 | NULL | | 47 | queuemem1 | queue6 | SIP/queuemem1 | 0 | NULL | | 48 | queuemem10 | queue6 | SIP/queuemem10 | 0 | NULL | | 49 | queuemem18 | queue6 | SIP/queuemem18 | 0 | NULL | | 50 | queuemem17 | queue6 | SIP/queuemem17 | 0 | NULL | | 51 | queuemem12 | queue6 | SIP/queuemem12 | 0 | NULL | | 52 | queuemem16 | queue6 | SIP/queuemem16 | 0 | NULL | | 53 | queuemem13 | queue6 | SIP/queuemem13 | 0 | NULL | +--++++-++ You can see that the member queuemem4 is first in line to be rang (has the first and lowest uniqueid in the table). But the first member that is being rang, is queuemem1. How come ?? Kind regards, Jonas. Jonas, We encountered the same problem. It is a bug in the Queue application. The Queue application actually orders members by their interface value. Here is the bug report I opened https://issues.asterisk.org/jira/browse/ASTERISK-18480 which was closed as Not A Bug by Digium. We worked around this by prepending an integer (001__, 002__, ...) to the interface in the database table and then removing it later in the dial plan. Hope this helps. Steven Wheeler -- _ -- 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
[asterisk-users] Problem/error with DAHDI tools 2.9.0.1
Hello, Getting this error on dahdi_cfg. Reverting to 2.8 the error goes away: *line 15: Unable to create 'dahdi_cfg' mutex.* Is this a problem? Thanks in advance. Full detail: [ebox] dahdi_cfg -vvv DAHDI Tools Version - 2.9.0.1 DAHDI Version: 2.9.0 Echo Canceller(s): HWEC Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 05) Channel 06: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 08) Channel 09: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 11) Channel 12: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 12) 12 channels to configure. Notice: Configuration file is /etc/dahdi/system.conf *line 15: Unable to create 'dahdi_cfg' mutex.* -- _ -- 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/error with DAHDI tools 2.9.0.1
On Wed, Feb 12, 2014 at 05:21:24PM +, Rodrigo Borges Pereira wrote: Hello, Getting this error on dahdi_cfg. Reverting to 2.8 the error goes away: *line 15: Unable to create 'dahdi_cfg' mutex.* Is this a problem? Thanks in advance. Full detail: [ebox] dahdi_cfg -vvv DAHDI Tools Version - 2.9.0.1 DAHDI Version: 2.9.0 Echo Canceller(s): HWEC Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 05) Channel 06: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 08) Channel 09: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 11) Channel 12: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 12) 12 channels to configure. Notice: Configuration file is /etc/dahdi/system.conf *line 15: Unable to create 'dahdi_cfg' mutex.* Which distro / version are you running? It appears that the sem_open call has failed on this platform. There does appear to be a mistake in the code though with the error reporting. 'perror' should be used instead of 'error'. The line number in the config file isn't related to this error report. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] Problem/error with DAHDI tools 2.9.0.1
Hello Shaun, This system is a custom distro, based on debian and currently built around kernel 2.6.35.8. Is sem_open introduced only in DAHDI 2.9 ? On Wed, Feb 12, 2014 at 5:37 PM, Shaun Ruffell sruff...@digium.com wrote: On Wed, Feb 12, 2014 at 05:21:24PM +, Rodrigo Borges Pereira wrote: Hello, Getting this error on dahdi_cfg. Reverting to 2.8 the error goes away: *line 15: Unable to create 'dahdi_cfg' mutex.* Is this a problem? Thanks in advance. Full detail: [ebox] dahdi_cfg -vvv DAHDI Tools Version - 2.9.0.1 DAHDI Version: 2.9.0 Echo Canceller(s): HWEC Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 05) Channel 06: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 08) Channel 09: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 11) Channel 12: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 12) 12 channels to configure. Notice: Configuration file is /etc/dahdi/system.conf *line 15: Unable to create 'dahdi_cfg' mutex.* Which distro / version are you running? It appears that the sem_open call has failed on this platform. There does appear to be a mistake in the code though with the error reporting. 'perror' should be used instead of 'error'. The line number in the config file isn't related to this error report. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
[asterisk-users] How does extensions.lua compares to extensions.conf ?
Hello, How does extensions.lua compares to extensions.conf or extensions.ael on stability, performance and features ? Would you recommand extensions.lua as an easy/easier way to access memcached, redis or equivalent ? Thoughs ? Comments ? Regards -- _ -- 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/error with DAHDI tools 2.9.0.1
Maybe the problem is lack of tmpfs? statfs(/dev/shm, 0xbff88568) = -1 ENOENT (No such file or directory) Is this a new requirement for DAHDI? On Wed, Feb 12, 2014 at 5:49 PM, Rodrigo Borges Pereira rodrigoborgespere...@gmail.com wrote: Hello Shaun, This system is a custom distro, based on debian and currently built around kernel 2.6.35.8. Is sem_open introduced only in DAHDI 2.9 ? On Wed, Feb 12, 2014 at 5:37 PM, Shaun Ruffell sruff...@digium.comwrote: On Wed, Feb 12, 2014 at 05:21:24PM +, Rodrigo Borges Pereira wrote: Hello, Getting this error on dahdi_cfg. Reverting to 2.8 the error goes away: *line 15: Unable to create 'dahdi_cfg' mutex.* Is this a problem? Thanks in advance. Full detail: [ebox] dahdi_cfg -vvv DAHDI Tools Version - 2.9.0.1 DAHDI Version: 2.9.0 Echo Canceller(s): HWEC Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 05) Channel 06: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 08) Channel 09: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 11) Channel 12: Hardware assisted D-channel (Default) (Echo Canceler: kb1) (Slaves: 12) 12 channels to configure. Notice: Configuration file is /etc/dahdi/system.conf *line 15: Unable to create 'dahdi_cfg' mutex.* Which distro / version are you running? It appears that the sem_open call has failed on this platform. There does appear to be a mistake in the code though with the error reporting. 'perror' should be used instead of 'error'. The line number in the config file isn't related to this error report. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] Problem/error with DAHDI tools 2.9.0.1
On Wed, Feb 12, 2014 at 05:49:49PM +, Rodrigo Borges Pereira wrote: Hello Shaun, This system is a custom distro, based on debian and currently built around kernel 2.6.35.8. Is sem_open introduced only in DAHDI 2.9 ? Yes. It was added in [1] in order to prevent errors when multiple invocations of dahdi_cfg are run in parallel. I've updated the error messages [2] on the master branch of dahdi-tools. That might provide some more information about why sem_open is failing on this platform. [1] http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=commit;h=9989b8779cef3 [2] http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=commit;h=066fa2aff33ba32 -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
[asterisk-users] Strange incoming call issue.
Hi all, I've got a customer who's reporting ghost calls. Essentially, the phone rings, they pick up, and there's no body there. It is NOT one-way audio, and it doesn't happen all the time. We use voipmonitor to watch calls, and this is what we saw for the call in question: | calldate| caller | called | duration | whohanged | +-++++-+ | 2014-02-12 09:28:06 | 575xxx | CCD539F38...-1 | 60 | NULL | | 2014-02-12 09:29:06 | 575xxx | CCD539F38...-2 |1 | NULL | So, it looks like my customer received a call, which lasted a minute, and then they hung up. Then their phone rang again, but there was no one there. Based on what I'm seeing in my log, the first call was never hung up, even though both parties claim to have hung up the call. My logs only indicate that the 'h' extension was called once, at 9:29:07 My question is, how can a call not get hung up when both parties hang up the call? I know that sounds odd, but that's what I'm seeing. Any ideas? Mike. -- _ -- 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/error with DAHDI tools 2.9.0.1
On Wed, Feb 12, 2014 at 05:55:41PM +, Rodrigo Borges Pereira wrote: Maybe the problem is lack of tmpfs? statfs(/dev/shm, 0xbff88568) = -1 ENOENT (No such file or directory) Is this a new requirement for DAHDI? tmpfs is not a requirement per-se, but it's how most POSIX libraries on linux (including the one installed on your system) implement the named semaphores. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] Problem/error with DAHDI tools 2.9.0.1
Well, I created /dev/shm and mounted tmpfs on it, and the problem is no more. Just creating /dev/shm was not enough. thanks. On Wed, Feb 12, 2014 at 6:06 PM, Shaun Ruffell sruff...@digium.com wrote: On Wed, Feb 12, 2014 at 05:55:41PM +, Rodrigo Borges Pereira wrote: Maybe the problem is lack of tmpfs? statfs(/dev/shm, 0xbff88568) = -1 ENOENT (No such file or directory) Is this a new requirement for DAHDI? tmpfs is not a requirement per-se, but it's how most POSIX libraries on linux (including the one installed on your system) implement the named semaphores. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] Problem/error with DAHDI tools 2.9.0.1
Just one last question: do you have another suggestion about this? thanks. On Wed, Feb 12, 2014 at 6:07 PM, Rodrigo Borges Pereira rodrigoborgespere...@gmail.com wrote: Well, I created /dev/shm and mounted tmpfs on it, and the problem is no more. Just creating /dev/shm was not enough. thanks. On Wed, Feb 12, 2014 at 6:06 PM, Shaun Ruffell sruff...@digium.comwrote: On Wed, Feb 12, 2014 at 05:55:41PM +, Rodrigo Borges Pereira wrote: Maybe the problem is lack of tmpfs? statfs(/dev/shm, 0xbff88568) = -1 ENOENT (No such file or directory) Is this a new requirement for DAHDI? tmpfs is not a requirement per-se, but it's how most POSIX libraries on linux (including the one installed on your system) implement the named semaphores. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] Problem/error with DAHDI tools 2.9.0.1
On Wed, Feb 12, 2014 at 06:12:41PM +, Rodrigo Borges Pereira wrote: Just one last question: do you have another suggestion about this? thanks. Not really. There are other ways that dahdi_cfg could serialize itself, but POSIX semaphores are widely deployed on systems that are installing newer versions of DAHDI. Although, dahdi-tools should probably have a configure script test for a working implementation of sem_open and use another mechanism if it is not available. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] Problem/error with DAHDI tools 2.9.0.1
Ok thanks Shaun! Best regards. On Wed, Feb 12, 2014 at 6:22 PM, Shaun Ruffell sruff...@digium.com wrote: On Wed, Feb 12, 2014 at 06:12:41PM +, Rodrigo Borges Pereira wrote: Just one last question: do you have another suggestion about this? thanks. Not really. There are other ways that dahdi_cfg could serialize itself, but POSIX semaphores are widely deployed on systems that are installing newer versions of DAHDI. Although, dahdi-tools should probably have a configure script test for a working implementation of sem_open and use another mechanism if it is not available. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
[asterisk-users] Asterisk Not Starting after YUM Update
Hi List, it feels silly, but here I am. My Asterisk box is useless, after running a long delayed yum update (Centos box). * A few details on the box: cat /etc/redhat-release CentOS release 5.10 (Final) arch i686 uname -a Linux hermes 2.6.18-371.4.1.el5 #1 SMP Thu Jan 30 06:09:24 EST 2014 i686 athlon i386 GNU/Linux asterisk -r Asterisk 1.6.2.20, Copyright (C) 1999 - 2010 Digium, Inc. and others. * Starting Asterisk very verbosely seems to load the dialplan, but at some point I get a segmentation fault. This is new to me! […] edited […] chan_agent.so = (Agent Proxy Channel) == Registered custom function 'EXTENSION_STATE' func_extstate.so = (Gets an extension's state in the dialplan) == Registered application 'DAHDIBarge' app_dahdibarge.so = (Barge in on DAHDI channel application) == Registered custom function 'CALLERPRES' == Registered custom function 'CALLERID' func_callerid.so = (Caller ID related dialplan functions) [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:760 load_module: G.729A transcoding module version 1.6.0_3.1.4, Copyright (C) 1999-2009 Digium, Inc. [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:761 load_module: This module is supplied under a commercial license granted by Digium, Inc. [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:762 load_module: Please see the full license text supplied by the accompanying [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:763 load_module: register utility, or ask for a copy from Digium. Segmentation fault The problem seems to come after the callerid module loads: does this make sense? BTW: I do have a G729 pack of licenses (they were actually active without any problem before messing with the update).. What should the clever sysadmin do? Thanks in advance, Aldo -- _ -- 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 Not Starting after YUM Update
On Wed, Feb 12, 2014 at 10:44:42PM +0100, Aldo Bergamini wrote: Hi List, it feels silly, but here I am. My Asterisk box is useless, after running a long delayed yum update (Centos box). [snip] Starting Asterisk very verbosely seems to load the dialplan, but at some point I get a segmentation fault. This is new to me! […] edited […] chan_agent.so = (Agent Proxy Channel) == Registered custom function 'EXTENSION_STATE' func_extstate.so = (Gets an extension's state in the dialplan) == Registered application 'DAHDIBarge' app_dahdibarge.so = (Barge in on DAHDI channel application) == Registered custom function 'CALLERPRES' == Registered custom function 'CALLERID' func_callerid.so = (Caller ID related dialplan functions) [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:760 load_module: G.729A transcoding module version 1.6.0_3.1.4, Copyright (C) 1999-2009 Digium, Inc. [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:761 load_module: This module is supplied under a commercial license granted by Digium, Inc. [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:762 load_module: Please see the full license text supplied by the accompanying [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:763 load_module: register utility, or ask for a copy from Digium. Segmentation fault The problem seems to come after the callerid module loads: does this make sense? BTW: I do have a G729 pack of licenses (they were actually active without any problem before messing with the update).. What should the clever sysadmin do? Thanks in advance, Aldo Try: # standard asterisk command-line. No verbosity strace -eopen asterisk -U asterisk -c See which module was the one last loaded. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.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] Asterisk Not Starting after YUM Update
DAHDI might be the culprit. You may have had a better version from Asterisk than the new one that YUM got you. Check to see if YUM gave you a new DAHDI. Who's your daddy now? You may want to rebuild the Asterisk DAHDI and install it over the DAHDI from your Linux distro. Ron On 12/02/2014 5:22 PM, Tzafrir Cohen wrote: On Wed, Feb 12, 2014 at 10:44:42PM +0100, Aldo Bergamini wrote: Hi List, it feels silly, but here I am. My Asterisk box is useless, after running a long delayed yum update (Centos box). [snip] Starting Asterisk very verbosely seems to load the dialplan, but at some point I get a segmentation fault. This is new to me! […] edited […] chan_agent.so = (Agent Proxy Channel) == Registered custom function 'EXTENSION_STATE' func_extstate.so = (Gets an extension's state in the dialplan) == Registered application 'DAHDIBarge' app_dahdibarge.so = (Barge in on DAHDI channel application) == Registered custom function 'CALLERPRES' == Registered custom function 'CALLERID' func_callerid.so = (Caller ID related dialplan functions) [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:760 load_module: G.729A transcoding module version 1.6.0_3.1.4, Copyright (C) 1999-2009 Digium, Inc. [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:761 load_module: This module is supplied under a commercial license granted by Digium, Inc. [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:762 load_module: Please see the full license text supplied by the accompanying [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:763 load_module: register utility, or ask for a copy from Digium. Segmentation fault The problem seems to come after the callerid module loads: does this make sense? BTW: I do have a G729 pack of licenses (they were actually active without any problem before messing with the update).. What should the clever sysadmin do? Thanks in advance, Aldo Try: # standard asterisk command-line. No verbosity strace -eopen asterisk -U asterisk -c See which module was the one last loaded. -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Gigaset R630H and Asterisk
Hi, Is anyone aware of an issue with Gigaset DECT handsets (R630H and N510P) and Asterisk? A client has them, and whenever they try a blind transfer, something goes wrong. Agent 1 starts and completes the blind transfer. Agent 2 answers the transferring call. Agent 2 hears asterisk music on hold, but the caller can hear the agent. Any ideas? Thanks Dan Journo Kesher Communications (UK) www.keshercommunications.comhttp://www.keshercommunications.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] Strange incoming call issue.
About a call not being hang up for asterisk while the client hang up, please remember SIP is based on UDP and UDP packets get easily lost... they are retransmitted but sometime they are lost as the previous... For the ghost calls, are the SIP port of the phones reachable from the Internet... maybe it is just someone trying to place some free calls Leandro 2014-02-12 19:05 GMT+01:00 Mike Diehl mdiehlena...@gmail.com: Hi all, I've got a customer who's reporting ghost calls. Essentially, the phone rings, they pick up, and there's no body there. It is NOT one-way audio, and it doesn't happen all the time. We use voipmonitor to watch calls, and this is what we saw for the call in question: | calldate| caller | called | duration | whohanged | +-++++-+ | 2014-02-12 09:28:06 | 575xxx | CCD539F38...-1 | 60 | NULL | | 2014-02-12 09:29:06 | 575xxx | CCD539F38...-2 |1 | NULL | So, it looks like my customer received a call, which lasted a minute, and then they hung up. Then their phone rang again, but there was no one there. Based on what I'm seeing in my log, the first call was never hung up, even though both parties claim to have hung up the call. My logs only indicate that the 'h' extension was called once, at 9:29:07 My question is, how can a call not get hung up when both parties hang up the call? I know that sounds odd, but that's what I'm seeing. Any ideas? Mike. -- _ -- 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] auto-answer call
Ø when i use the Dial the sip/105 still ringing This should help you out http://wiki.snom.com/FAQ/How_to_make_Asterisk_send_INVITEs_to_trigger_the_phone_for_Intercom Dan Journo Kesher Communications (UK) www.keshercommunications.comhttp://www.keshercommunications.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
[asterisk-users] how to selectively disable callerid block?
In Asterisk 1.8, I used the following line in extensions.conf to allow
me to pass *82 in front of a dialed number, to disable the callerid
block that's normally on that POTS line:
; disable callerid block
exten = _*82.,1,Dial(${POTS}/${EXTEN})
But this seems to have stopped working when I upgraded to Asterisk
11.7. I get the following debug output, with a no call pickup
possible message as soon as I press the '8':
[Feb 12 18:17:39] -- Starting simple switch on 'DAHDI/2-1'
[Feb 12 18:17:39] DEBUG[2339]: devicestate.c:442 devstate_event: device
'DAHDI/2' state '2'
[Feb 12 18:17:42] DEBUG[5898][C-000e]: sig_analog.c:1600
analog_handle_dtmf: Begin DTMF digit: 0x2A '*' on DAHDI/2-1
[Feb 12 18:17:42] DEBUG[5898][C-000e]: chan_dahdi.c:2145 my_handle_dtmf:
Begin DTMF digit: 0x2A '*' on DAHDI/2-1
[Feb 12 18:17:42] DEBUG[5898][C-000e]: sig_analog.c:1600
analog_handle_dtmf: End DTMF digit: 0x2A '*' on DAHDI/2-1
[Feb 12 18:17:42] DEBUG[5898][C-000e]: chan_dahdi.c:2145 my_handle_dtmf:
End DTMF digit: 0x2A '*' on DAHDI/2-1
[Feb 12 18:17:42] DEBUG[5898][C-000e]: sig_analog.c:2121
__analog_ss_thread: waitfordigit returned '*' (42), timeout = 0
[Feb 12 18:17:44] DEBUG[5898][C-000e]: sig_analog.c:1600
analog_handle_dtmf: Begin DTMF digit: 0x38 '8' on DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: chan_dahdi.c:2145 my_handle_dtmf:
Begin DTMF digit: 0x38 '8' on DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: sig_analog.c:1600
analog_handle_dtmf: End DTMF digit: 0x38 '8' on DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: chan_dahdi.c:2145 my_handle_dtmf:
End DTMF digit: 0x38 '8' on DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: sig_analog.c:2121
__analog_ss_thread: waitfordigit returned '8' (56), timeout = 0
[Feb 12 18:17:44] DEBUG[5898][C-000e]: chan_dahdi.c:5075 dahdi_enable_ec:
Enabled echo cancellation on channel 2
[Feb 12 18:17:44] DEBUG[5898][C-000e]: features.c:7880 ast_pickup_call:
pickup attempt by DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: features.c:7900 ast_pickup_call: No
call pickup possible... for DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: sig_analog.c:2211
__analog_ss_thread: No call pickup possible...
and then a busy signal.
I didn't pay much attention to the differences between Asterisk 1.8
and 11.7 since everything seemed to still work ... Can someone point
me in the right direction?
--
Eric Cooper e c c @ c m u . e d u
--
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Re: [asterisk-users] How does extensions.lua compares to extensions.conf ?
On Wed, Feb 12, 2014 at 12:50 PM, Olivier oza.4...@gmail.com wrote: Hello, How does extensions.lua compares to extensions.conf or extensions.ael on stability, performance and features ? Would you recommand extensions.lua as an easy/easier way to access memcached, redis or equivalent ? Thoughs ? Comments ? The lack of replies should give you your answer. Extensions AEL and LUA don't get much action these days, I'm sure there are a few people that use them but extensions.conf has way more code coverage from a testing POV. Your better off using AGI if you want to leverage redis or memcached. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 does extensions.lua compares to extensions.conf ?
On Wed, Feb 12, 2014 at 6:26 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Feb 12, 2014 at 12:50 PM, Olivier oza.4...@gmail.com wrote: Hello, How does extensions.lua compares to extensions.conf or extensions.ael on stability, performance and features ? Would you recommand extensions.lua as an easy/easier way to access memcached, redis or equivalent ? Thoughs ? Comments ? The lack of replies should give you your answer. Extensions AEL and LUA don't get much action these days, I'm sure there are a few people that use them but extensions.conf has way more code coverage from a testing POV. Your better off using AGI if you want to leverage redis or memcached. Actually, I use Lua dialplans in several production systems. Some are used in conjunction with traditional dialplans and some are the only source of dialplans. They've always been rock solid. I actually find it easier to configure even a moderately complex dialplan than the traditional dialplan syntax. -- _ -- 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 V10, SIP MESSAGE fails for unknown reason, missing DNS-lookup?
Hi, I'm using SIP MESSAGE to asterisk V10 and it fails to be received. I'm not super sure of the reason but I'm making this guess: Due to I'm using non ipaddress in the to field, which contains sip:mobil1.xyz.com, Asterisk makes the mistake to try matching this name mobil1.testserver.com in extensions.conf and no extension/peer is found in the sip-message context I've configured. It works when the TO: field contains an numeric ipadress. Any hints how to be able to make asterisk first look up the ip before trying to match it? /Johan LOG [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c: --- SIP read from UDP:83.186.238.111:5060 --- MESSAGE sip:mobil1.xyz.com SIP/2.0 Via: SIP/2.0/UDP 83.186.238.111:5060;branch=z9hG4bK-3f138a53 To: sip:mobil1.xyz.com From: sip:83.186.238.111;tag=7a82b127 Call-ID: 857d4ed8@83.186.238.111mailto:857d4ed8@83.186.238.111 CSeq: 245 MESSAGE Max-Forwards: 70 User-Agent: CareIP 7813409 v1.2.4.0 Content-Type: application/scaip+xml Content-Length: 138 My message - [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 0 [ 49]: MESSAGE sip:mobil1.xyz.com SIP/2.0 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 83.186.238.111:5060;branch=z9hG4bK-3f138a53 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 2 [ 39]: To: sip:mobil1.xyz.com [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 3 [ 39]: From: sip:83.186.238.111;tag=7a82b127 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 4 [ 32]: Call-ID: 857d4ed8@83.186.238.111mailto:857d4ed8@83.186.238.111 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 5 [ 17]: CSeq: 245 MESSAGE [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 7 [ 35]: User-Agent: CareIP 7813409 v1.2.4.0 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 8 [ 35]: Content-Type: application/scaip+xml [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 9 [ 19]: Content-Length: 138 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 10 [ 0]: [Feb 12 15:13:59] DEBUG[25824] chan_sip.c:Body 0 [138]: My message [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c: --- (10 headers 1 lines) --- [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: = Looking for Call ID: 857d4ed8@83.186.238.111mailto:857d4ed8@83.186.238.111 (Checking From) --From tag 7a82b127 --To-tag [Feb 12 15:13:59] DEBUG[25824] acl.c: For destination '83.186.238.111', our source address is '172.26.19.13'. [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Target address 83.186.238.111:5060 is not local, substituting externaddr [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.105.99.108:5060 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Allocating new SIP dialog for 857d4ed8@83.186.238.111mailto:857d4ed8@83.186.238.111 - MESSAGE (No RTP) [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c: No matching peer for '83.186.238.111' from '83.186.238.111:5060' [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c: Looking for s in sipmessage (domain mobil1.xyz.com) [Feb 12 15:13:59] WARNING[25812] pbx.c: Channel 'Message/ast_msg_queue' sent into invalid extension 'mobil1.xyz.com' in context 'sipmessage', but no invalid handler Johan Sandgren Software Engineer Svep Design Center AB S:t Lars väg 42A 222 70 Lund, Sweden Phone +46 46 192 722 E-mail j...@svep.semailto:j...@svep.se Website www.svep.sehttp://www.svep.se -- _ -- 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] Gigaset R630H and Asterisk
Since there is no Transfer button for SIP INVITEs, I guess it is a DTMF related problem. At first I would check whether the ways of transmitting and receiving DTMF signals are compatible (http://www.voip-info.org/wiki/view/Asterisk+DTMF). I have a similar setup where there are no problems with mobile DECT devices. jg -- _ -- 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
