Re: [asterisk-users] Are there AMI commands to manipulate a voice mailbox?
On 14/05/2010, Motiejus Jakštys desired@gmail.com wrote: Talking about file permissions, on Linux everything is possible using POSIX ACLs. You can set specific rights to files/directories for certain users. Note 1: if setting group permissions is enough, use that. Note 2: Asterisk and web server should be on separate machines (at least virtual machines) for many reasons... I would mount my voicemail over NFS... which itself has enough access control. Note 3: if you decide to experiment with ACLs (IMHO, most flexible) - do not forget to remout your file system: mount -o remount,acl /usr Not quite everything - you're still limited to read/write/execute granularity (unless something has changed in the 5 years since I experimented with it). If you're expecting Full Control / Modify / Delete etc as per Windows 2000 and its ilk, you might have to look at something else... Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett -- _ -- 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] app_addon_sql_mysql.c:116 find_identifier
The voip-info.org tells it differently :
*MYSQL(Clear* ${resultid}*)*
Frees memory and data structures associated with result set.
*MYSQL(Disconnect* ${connid}*)*
Disconnects from named connection to MySQL.
But it does not make a difference...
It's strange that every mysql-query is the same :
[May 14 09:20:24] -- Executing [...@macro-queryastdb:5]
MYSQL(Local/2...@105002-ca25,2, Clear 20) in new stack
[May 14 09:20:24] -- Executing [...@macro-queryastdb:6]
MYSQL(Local/2...@105002-ca25,2, Disconnect 19) in new stack
[May 14 09:20:24] -- Executing [...@macro-getsipaccount:5]
MYSQL(Local/2...@105002-ca25,2, Clear 20) in new stack
[May 14 09:20:24] -- Executing [...@macro-getsipaccount:6]
MYSQL(Local/2...@105002-ca25,2, Disconnect 19) in new stack
[May 14 09:20:34] -- Executing [...@macro-getmailboxfromsipuserid:5]
MYSQL(SIP/testcorp-083a2128, Disconnect 19) in new stack
[May 14 09:20:34] -- Executing [...@macro-getmailboxfromsipuserid:6]
MYSQL(SIP/testcorp-083a2128, Clear 20) in new stack
And then the last MySql-query :
[May 14 09:20:40] -- Executing [...@macro-sdgeenopname:5]
MYSQL(SIP/testcorp-083a2128, Clear 20) in new stack
[May 14 09:20:40] WARNING[5939]: app_addon_sql_mysql.c:116
find_identifier: Identifier 20, identifier_type 2 not found in
identifier list
[May 14 09:20:40] WARNING[5939]: app_addon_sql_mysql.c:355 aMYSQL_clear:
Invalid result identifier 20 passed in aMYSQL_clear
[May 14 09:20:40] -- Executing [...@macro-sdgeenopname:6]
MYSQL(SIP/testcorp-083a2128, Disconnect 19) in new stack
The only difference is that this is a macro which is executed in the
h-extension (so after the channel has been hung up). Could this be the
issue ??
If so, what is the solution ?? No real solution is needed as it does not
mess up anything, but it still is a WARNING.
Jonas.
On 05/13/2010 04:26 PM, Doug Lytle wrote:
Jonas Kellens wrote:
exten = s,n,NoOp(fetchid = ${fetchid})
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
The only different between yours and mine is that I do a disconnect
before I do the clear.
Try:
exten = s,n,MYSQL(Fetch fetchid ${resultid} extensie)
exten = s,n,NoOp(fetchid = ${fetchid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,MYSQL(Clear ${resultid})
My guess is (If it works), that you can't clear an open channel.
Doug
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Re: [asterisk-users] No ringtone when going from queue to dial-command
The only thing that makes it ring in step 3 (so after the queue) is calling the Queue-command with the r-option. So there is no music on hold but a ringtone, when the caller sits in the queue. Now the question is: when I want to use music on hold while inside the queue, how can I get the ringtone back ?! Jonas. On 05/12/2010 05:37 PM, Jonas Kellens wrote: Yes, 20 in Queue is timeout... works fine. Also with the Ringing() command, there is no dialtone... It's just silence... With or without the r-option, always the same. When there is no Queue in between the 2 dial-commands, then the ringtone is there as it should be ! So when I change to the Queue and to musiconhold, I loose the ringtone... Should I do something after the Queue-command to get the ringing back ?? Ringing() does not help in my case... Jonas. On 05/12/2010 12:03 PM, Vardan wrote: Try so: 1. dial(SIP/account1,20) 2. queue(myqueue,,,20) 3. Ringing 4. dial(SIP/account2,,r) 20 in queue is timeout? http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Vardan -- _ -- 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] is my PHPAGI Soap code right?
Hello, i try to use soap in the phpagi. i want to call a function from a web service when a user call a telephne failed. this is my phpagi script, Could you show me what's wrong ? becasue i can't excute it successfully. please open the following url to see my code: http://pastebin.com/uzvWSxPy Thanks! -- Thanks for your supporting, have a nice day. Sucan -- _ -- 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] app_addon_sql_mysql.c:116 find_identifier
Jonas Kellens wrote: But it does not make a difference... I'm running Asterisk 1.4.x, what version are you running? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 between two Asterisk servers [SOLVED]
Hi! Issue solved. Looks like all I was missing was one parameter: fromuser= That's interesting - could be related to this: http://lists.digium.com/pipermail/asterisk-dev/2006-November/024842.html You were probably caught be the fact that you are using extension numbers also as SIP user names for your phones (here: 3666). This is not a good thing to do, better use an alphanumeric username or the phone's MAC address etc. As for your IAX sound quality issue: I have seen that before as well, and switched to SIP (as others did). My guess is that it will probably go away if you use Asterisk 1.4 on both sides, though. SIP DEBUG on the receiving Asterisk gives you a hint which peer was found if matching is done on the IP address, the text is somethint like Found peer ... or Found no matching peer or user for w.x.y.z Two quotes from the Wiki to explain things better: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer When Asterisk receives an incoming SIP call, the SIP Channel Module 1. first tries to find a [user] section matching the caller name (From: username), 2. then tries to find a [peer] section matching the caller's IP address. 3. If no matching user or peer is found, the call is sent to the context defined in the [general] section of sip.conf. As of Asterisk 1.2, there is no reason to actually use 'user' entries any more at all; you can use 'type=peer' for everything and the behavior will be much more consistent. All configuration options supported under 'type=user' are also supported under 'type=peer'. The difference between friend and peer is the same as defining _both_ a user and peer, since that is what 'type=friend' does internally. The only benefit of type=user is when you _want_ to match on username regardless of IP the calls originate from. If the peer is registering to you, you don't need it. If they are on a fixed IP, you don't need it. 'type=peer' is _never_ matched on username for incoming calls, only matched on IP address/port number (unless you use insecure=port or higher). Philipp -- _ -- 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] app_addon_sql_mysql.c:116 find_identifier
It does not make a difference because it is the same result : All the other queries go well, just the last one gives this 'WARNING'. Using 1.4.25.1 Jonas. On 05/14/2010 11:46 AM, Doug Lytle wrote: Jonas Kellens wrote: But it does not make a difference... I'm running Asterisk 1.4.x, what version are you running? Doug -- _ -- 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 my PHPAGI Soap code right?
- Original Message - Hello, i try to use soap in the phpagi. i want to call a function from a web service when a user call a telephne failed. this is my phpagi script, Could you show me what's wrong ? becasue i can't excute it successfully. please open the following url to see my code: http://pastebin.com/uzvWSxPy Thanks! Perhaps if you explained what errors you were seeing would help ? Have you tried running it from the CLI to see if the syntax is correct ? -- Thanks, Phil -- _ -- 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] app_addon_sql_mysql.c:116 find_identifier
Jonas Kellens wrote: It does not make a difference because it is the same result : All the other queries go well, just the last one gives this 'WARNING'. You may want to give func_odbc a try, several say it's a better way: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+func_odbc Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Agents
I've been trying to get the hang of Agents and Queues and I must say
its a little unclear as to how things work.
So maybe someone has some better idea
From what I can work out an Agent is meant to be nothing more than a
virtual device that can be moved around physical devices... by login
and logging out. Queues can contain any type of interface not a point
that is partially well put in the Sark we have just got nore in the
voi-info website It also seams to suggest that Agents are a
deprecated feature.
AgentLogin.
AgentCallbackLogin is depreciated but what has it been replaced by?
Not sure what AgentLogin is actually useful for.
AgentCallbackLogin in the Management API does not set
${AGENTBYCALLERID_${CALLERID(num)}} I guessing this is a error,
fortunatly I've worked out a way to get round it. (setvar)
The is no way to log an agent in from the Command Line Interface.
AgentLogoff
Easy so long as you know the agent id you need to logoff, which means
using ${AGENTBYCALLID_${CALLERID(num)}}
Queues really have very little to do with Agents as any type of device
can be statically on a queue or dynamically added when needed, but the
info I've found seams to heavily tie the two concepts together.
Peter
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Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
--- On Thu, 5/13/10, Zoa zoach...@securax.org wrote: Can you try trunk = no ? Lifesaver... trunk=no made the interference go away. I have clean audio now. Quote: IAX Trunking needs support of a hardware timer. I'm supposing my system is using the DAHDI-driven Digium cards on my motherboard. I don't know how hardware timers work and if Digium hardware rely on the motherboard (my system clock is going too fast and my ntpd is constantly adjusting the clock by -2.6 seconds every 20 minutes). In any case, since I'm on a dedicated LAN I guess I can safely set trunk=no. Thanks! Vieri -- _ -- 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 with ringinuse=no, queue members receive randomly two calls
i've also tied this tests: - changed hardware - upgrade to 1.4.31 - kernel recompiled with 1000 Hz option - changed SO (Slackware 13) - run the system on hardware (no ESXi) But i've not resolved the problem. Do you have any idea? On Thu, May 6, 2010 at 11:54 AM, nik600 nik...@gmail.com wrote: i get may debug messages like this: DEBUG[30684] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=-1) Is because dahdi is not installed? Can this be a possible cause of this behaviour? -- /*/ nik600 http://www.kumbe.it -- _ -- 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 between two Asterisk servers [SOLVED]
--- On Fri, 5/14/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: You were probably caught be the fact that you are using extension numbers also as SIP user names for your phones (here: 3666). This is not a good thing to do, better use an alphanumeric username or the phone's MAC address etc. Is there more info on this? I mean, why is it bad, apart from the security implication. As for your IAX sound quality issue: I have seen that before as well, and switched to SIP (as others did). My guess is that it will probably go away if you use Asterisk 1.4 on both sides, though. It went away even with 1.2 but I needed to set trunk=no. Probably a jitter buffer issue on my system(s). SIP DEBUG on the receiving Asterisk gives you a hint which peer was found if matching is done on the IP address, the text is somethint like Found peer ... or Found no matching peer or user for w.x.y.z Tnanks for the info Philipp. I'll try to further debug my SIP messages. Vieri -- _ -- 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] bad magic number log messages
The way I reproduce it is not simple. I am receiving calls on 1.6.0.27 and doing FastAGI scripting. Then it happens after a few hours of steady calls. I'm looking for a simpler way to reproduce it. Looks like issue 0017321 describes the bug in one way. This is a serious bug, I am checking to see if it is in 1.6.2.7 since 1.6.0.x is just for security fixes now apparently. I have found that 1.6.2 is more unstable for some reason than 1.6.0 at ~180 calls on my system. Not sure why... So I'm trying to stick to 1.6.0. John From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alec Davis Sent: Wednesday, May 12, 2010 2:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: 'Asterisk Developers Mailing List' Subject: Re: [asterisk-users] bad magic number log messages I should have added, that if you havn't already, please report your senario with example dialplan etc to one of the open bug reports related to you problem, otherwise feel free to open a new one. Also 'many' was a bit strong, should have said 'others'. Alec Davis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alec Davis Sent: Thursday, 13 May 2010 7:52 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] bad magic number log messages Many are having this problem. goto http://issues.asterisk.org and search for 'bad magic number' Notably, a few reports have come up in recent days. Alec Davis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Rose Sent: Thursday, 13 May 2010 3:00 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] bad magic number log messages Anyone else get this issue - around 200 entries per second of this in the Asterisk messages file: astobj2.c:115 INTERNAL_OBJ: bad magic number 0x27b4113a Seems to happen after several hours of receiving a steady stream of test calls. My messages file is 7.5 gigs... John -- _ -- 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] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
I think that the clock resets would cause no audio or garbled audio every 20 minutes, not constant interference. Could you tell us how many simultaneous calls were in the trunk and what the size is of 1 voice packet ? Can you try putting maximum 30 calls per trunk (use multiple trunks if needed) and see if the problem goes away. Greetings, zOa Vieri wrote: --- On Thu, 5/13/10, Zoa zoach...@securax.org wrote: Can you try trunk = no ? Lifesaver... trunk=no made the interference go away. I have clean audio now. Quote: IAX Trunking needs support of a hardware timer. I'm supposing my system is using the DAHDI-driven Digium cards on my motherboard. I don't know how hardware timers work and if Digium hardware rely on the motherboard (my system clock is going too fast and my ntpd is constantly adjusting the clock by -2.6 seconds every 20 minutes). In any case, since I'm on a dedicated LAN I guess I can safely set trunk=no. Thanks! Vieri -- _ -- 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 and codec negotiation
Hi, If I am expecting too much here, please just tell me so, but I was under the impression that this was put into 1.6.x I have 2 types of SIP devices. For argument's sake, let us say that one type of device can talk G722 and ALAW, and the other only talks ALAW. I have directmedia=yes. Calls originated from ALAW only devices work great. Calls from G722 to G722 devices work great. ...but the G722 to ALAW calls do not work. I can see from the SIP trace that this is because Asterisk makes no attempt to modify the codecs in its directmedia re-INVITE packets to ensure that the 2 parties can talk, so you end up with an asymmetric codec stream between the handsets, which results in silence both ways. I would expect Asterisk to either determine that there are no common codecs, and do an implicit directmedia=no for the remainder of the call, or to only send the list of common codecs to each party in the re-INVITE's SDP (There is a room for a per-device can_change_codec=bool parameter in there too I think). For 1.4 there was a popular codec negotiation patch which I believe fixed this. Is this not in 1.6? Am I missing something else perhaps? Thanks for any pointers. Regards, Steve -- _ -- 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] Need fax solution for 1.4.xx (Resolution)
On 05/13/2010 10:48 PM, William Stillwell (Lists) wrote: Ok, I ended up upgrading 2 of my 5 boxes to 1.6.2.7 , and using spandsp 0.0.6pre17, dahdi-linux-complete-2.3.0+2.3.0 , and enabled app_fax. Hint: you need to install spandsp then run ./configure then make menuselect :) I was able to send over a 50 page fax from coast to coast with 0 issues However, did get this message in CLI: [May 13 07:27:31] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:27:31] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:14] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:14] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:27] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:27] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:39] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:39] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found However, there was no noticeable errors in the fax., googling, the error didn't seem to make much since. This was via copper pair, over traditional LD carrier, into PRI terminating into a Sangoma card. Intel Xeon x3460, 8 gb ram, 320gb raid 0 sata Thanks to all who offered suggestions, and such, I will try this out, and hopefully should work well, as Steve Hinted to a year ago. William Stillwell Those messages mean exactly what they say - a chunk of image data was not decoded by the modem. The FAX protocol will retry the missing chunk of image, and by the end of the FAX you probably see no problems at all. The FAX will, however, taking somewhat longer than it should. Regards, Steve -- _ -- 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 queues membername problem
Hi, I'm using dynamic realtime with asterisk 1.6.0.24, I'm having a strange problem with queue_members... If I update only 'membername' field on queue_members table asterisk won't refresh the change, but if I update another field like interface everything works as expected, I've found this problem also deleting a existing agent on queue_members and then inserting a new one with the same interface, penalty and pause but with another membername :( Asterisk won't refresh the change and show the old membername on CLI (queue show my-queue...). It is possible that asterisk refresh these info? 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] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
On Fri, 14 May 2010, Vieri wrote: I'm supposing my system is using the DAHDI-driven Digium cards on my motherboard. I don't know how hardware timers work and if Digium hardware rely on the motherboard (my system clock is going too fast and my ntpd is constantly adjusting the clock by -2.6 seconds every 20 minutes). In any case, since I'm on a dedicated LAN I guess I can safely set trunk=no. Maybe it's just me, but I'd be thinking if the mobo manufacturer did such a crappy job on the clock, what else is wrong. I'd be looking for a better mobo. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] 1.6.2.7 SIP realtime problem
I'm getting the following message in my full log at startup and my realtime sip peers aren't being found. My realtime extensions have no errors. The table sippeers exists in the database. Is this a known problem? res_config_mysql.c: Table sippeers not found in database. This table should exist if you're using realtime. -- _ -- 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] Need fax solution for 1.4.xx (Resolution)
Steve, Thanks for the heads up, after extensive testing today, I have decided to edit t30.c to mark as debug, as I recorded a call between two analog fax machines using mixmonitor, and noticed the same waveform patterns, as a call into spandsp/receivefax. I must admit, I am way happier with spandsp/receivefax/asterisk 1.6.2.7 then asterisk 1.4.x w/pikafax. Keep up the good work :) William Stillwell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Underwood Sent: Friday, May 14, 2010 12:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Need fax solution for 1.4.xx (Resolution) On 05/13/2010 10:48 PM, William Stillwell (Lists) wrote: Ok, I ended up upgrading 2 of my 5 boxes to 1.6.2.7 , and using spandsp 0.0.6pre17, dahdi-linux-complete-2.3.0+2.3.0 , and enabled app_fax. Hint: you need to install spandsp then run ./configure then make menuselect :) I was able to send over a 50 page fax from coast to coast with 0 issues However, did get this message in CLI: [May 13 07:27:31] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:27:31] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:14] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:14] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:27] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:27] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:39] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:39] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found However, there was no noticeable errors in the fax., googling, the error didn't seem to make much since. This was via copper pair, over traditional LD carrier, into PRI terminating into a Sangoma card. Intel Xeon x3460, 8 gb ram, 320gb raid 0 sata Thanks to all who offered suggestions, and such, I will try this out, and hopefully should work well, as Steve Hinted to a year ago. William Stillwell Those messages mean exactly what they say - a chunk of image data was not decoded by the modem. The FAX protocol will retry the missing chunk of image, and by the end of the FAX you probably see no problems at all. The FAX will, however, taking somewhat longer than it should. Regards, Steve -- _ -- 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] 1.6.2.7 SIP realtime problem
On Friday 14 May 2010 16:09:58 Bruce Ferrell wrote: I'm getting the following message in my full log at startup and my realtime sip peers aren't being found. My realtime extensions have no errors. The table sippeers exists in the database. Is this a known problem? res_config_mysql.c: Table sippeers not found in database. This table should exist if you're using realtime. Check your [context] in res_mysql.conf. In previous versions, it was set as [general], but extconfig.conf had asterisk as the name of the connection. These two configuration files need to match. It's correct in the sample configs, but if you upgraded from a prior version, it's possible that you still have the bad match. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) 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] Problem with Music on hold
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! During tests with a Grandstream GXP280 phone, I found that pressing 'hold' button, the other extension (Qutecom softphone) is put on hold but without music. Then, when resuming the conversation, I listen the other user again but he/her no longer listen to me. When from softphone the same test is realised, it does not happen this problem. Can it be due to a configuration problem of the Grandstream phone? Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvuBdQACgkQZpa/GxTmHTdC8QCgilVwAkPv7VIS6AOA7pzGKmgQ EYcAnRxcvcFHAAwsVZ+Vg2ukWoPSWzmL =xdok -END PGP 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] 1.6.2.7 SIP realtime problem
On 05/14/2010 04:17 PM, Tilghman Lesher wrote: On Friday 14 May 2010 16:09:58 Bruce Ferrell wrote: I'm getting the following message in my full log at startup and my realtime sip peers aren't being found. My realtime extensions have no errors. The table sippeers exists in the database. Is this a known problem? res_config_mysql.c: Table sippeers not found in database. This table should exist if you're using realtime. Check your [context] in res_mysql.conf. In previous versions, it was set as [general], but extconfig.conf had asterisk as the name of the connection. These two configuration files need to match. It's correct in the sample configs, but if you upgraded from a prior version, it's possible that you still have the bad match. That did it. in res_mysql it's [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = astuser dbpass = astpass dbport = 3306 dbsock = /var/run/mysql/mysql.sock requirements=warn in extconfig the line sippeers = mysql,asterisk,sip_buddies asterisk pointed to the dbname in res_mysql, not the context. It still works that way in 1.4.31 That was fun... NOT! :) -- _ -- 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
