[asterisk-users] Asterisk takes incoming call before extension was submitted
I have a Problem with incoming ISDN calls in Austria, I use zaptel and asterisk bristuffed from Debian/Etch. - If someone outside is dialing the phonenumber and the extension on an ISDN phone, asterisk catches the call and puts into in the s extension before the extension was submittet. - The same Number dialed from a mobilephone works as expected, asterisk recieves the extension. As well with stored numbers in ISDN phones. From Diskussions in german Forums it turns out, that Austria does not follow the DSS1 Standard (what ever this is...) and there is a timing Problem with the zapata ISDN configuration. Does anyone know, how I can increase the timout value for the ISDN implses, so asterisk waits for the extension on the ISDN channel? Regards, martin My zapata.conf: [trunkgroups] [channels] language=de pridialplan=local prilocaldialplan=local nationalprefix = 00 internationalprefix = 000 ; trust user provided callerid (clip no screening)? pritrustusercid = yes ; hidecallerid=no callerid=asreceived switchtype = euroisdn signalling = bri_cpe pridialplan = local prilocaldialplan = local echocancel = yes echocancelwhenbridged=no echotraining=no usecallerid = yes overlapdial = yes immediate = no group = 1 context = isdn channel = 1-2 switchtype = euroisdn signalling = bri_cpe pridialplan = local prilocaldialplan = local echocancel = yes echocancelwhenbridged=no echotraining=no usecallerid = yes overlapdial = yes immediate = no group = 1 context = isdn channel = 4-5 echotraining = yes rxgain = 0.8 txgain = 0.8 signalling = fxo_ks context = default channel = 7 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Penalties not working properly
I am using asterisk 1.4.18. I cant at this stage upgrade to any latest version. Linear strategy for queues is not in asterisk 1.4.18. I have to use ringall instead. Is it possible Disabling call-waiting for my agents only?? While other sip users have call waiting functionality. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robin Rodriguez Sent: Tuesday, August 05, 2008 11:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly Syed Nasruddin wrote: Hi, Actully the way I want the penalties functionality to behave it is not doing it accordingly. I am right now using ringall. Set penalty 1 for one agent and 2 for secnd agent. All the calls come in and go to first agent#1 having penalty one. But the second call also go to agent#1 and start waiting for it to be free rather it should have gone to penalty two agent#2 I have added call-limit=1 for bot sip accounts. And started the services. Still find the status wrong. nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: Tuesday, August 05, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 5:27 PM, Syed Nasruddin [EMAIL PROTECTED] wrote: Cannot i use ringall strategy with penalties??? Will rrmemory will fullfil my requirement?? rrmemory isn't ringall, it won't ring all members. But yes - you can use ringall with penalties. My requirements: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 3. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Moreover why my queue status shows my agent as NOT IN USE while in fact it is busy answering the call?? What you are seeing is caused by status NOT IN USE. You have to set call-limit in sip.conf for all your phones, to any value, so that device states work correctly, and queue can know that those phones are busy. Now you probably can see in CLI that queue is sending second call to first agent(s). Regards, Atis Thanks Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, August 05, 2008 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 Robin, round robin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Very carefully reread the descriptions on penalties and queue strategies on voip-info.org, the first time I
Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 13
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will be on vacation until Tuesday 19th of August with limited access to voice and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on my return. Dimitri Osler ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] does astcanary really work?
A week ago, I tried give realtime priority to asterisk proces using -p switch, asterisk was running inside astcanary, but yestarday asterisk probably starts eating all cpu and lock any access to computer, only ping was possible, so, anybody have experience, that ascanary process does really work to lower process priority in case of overloading? PJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream RS-232 config (slightly off-topic)
--- On Tue, 8/5/08, Atis Lezdins [EMAIL PROTECTED] wrote: Have you tried powering it on, while holding reset button? Yes, several times. I contacted Grandstream's helpdesk and they told me to keep the reset button pressed while I plug the power off and back on again. I even tried keeping it pressed for 5 minutes and the device kept rebooting but the factory defaults were never restored. Additionally you can try to leave it for week powered off and hope that there's some old battery keeping up settings. I'll try that... :-( Are you sure that there isn't some enable admin mode command in telnet? It should allow you everything that's available from web. I wish there were but I haven't found anything. I even tried enable admin mode (just in case this was my lucky day) but it doesn't compute... I suspect that the built-in HTTP server crashes (which would be a GS bug) as soon as the device boots maybe because of a problem with the configuration. Since I can't reset to defaults I tried to setup a default config file (cfgMAC file) on my LAN HTTP provisioning server (whose URL I had previously setup in the faulty device). When the GS device boots I can see from my Apache log that the cfgMAC file is being fetched OK. However, the device's configuration is left unchanged. So it loads the cfgMAC but it doesn't commit it/store it or something. Something's definitely wrong there. Curiously, I checked two other GXW4008 on my LAN (both are working fine). However, if I connect to both via telnet, I can see that one of them has a reset to factory defaults command and the other doesn't (just like the faulty device). As you can see here the settings on both devices are identical: ATA #13: Grandstream GXW-4008 V1.3A Command Shell Copyright 2006-2008 Supported commands: config -- Configure the device status -- Show device status upgrade -- Upgrade the device reboot -- Reboot the device reset -- Factory reset help-- Show this help text exit-- Exit this command shell Software Versions: Main -- 1.0.0.86 Boot -- 1.0.0.7 Core -- 1.0.0.21 Base -- 1.0.0.60 ATA #2: Grandstream GXW-4008 V1.3A Command Shell Copyright 2006-2008 Supported commands: config -- Configure the device status -- Show device status upgrade -- Upgrade the device reboot -- Reboot the device help-- Show this help text exit-- Exit this command shell Software Versions: Main -- 1.0.0.86 Boot -- 1.0.0.7 Core -- 1.0.0.21 Base -- 1.0.0.60 Really odd (both systems have customized configs). Anyway, thanks for your feedback. I hope GS will come up with a new firmware with a complete list of telnet commands. Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] shared mysql connection in dialplan
hi all, i just finished developing some incoming call features in a macro. that macro gets executed everytime an incoming call is received and a new mysql connection is made using the MYSQL cmd in dialplan. i want to use a single mysql connection for every incoming call. my idea of doing it is like this, i want to get a mysql connection in a global variable, just to share the connection with different incoming calls. Im not sure if this can be done. I am going to try doing it somehow, meanwhile i want your suggestions about how i can share a mysql connection with different calls in a dialplan. I am using asterisk1.4.2 and asterisk addon1.4.0 package for mysql connectivity. Thanx in advance -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Avaya
Good question, I'll check. Regards, Steve 2008/8/6 Tom Lynn [EMAIL PROTECTED]: Steve, what kind of Avaya system is this? They make several. On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote: Hi, Sorry this is so long, but I am reasonably desparate. I am having real fun with hooking an Avaya system to Asterisk using ISDN30. I have the ISDN signalling all sorted one way, and can pass calls from the real world (ie. the telco and asterisk) TO the avaya box, and it accepts that and sets up the call perfectly. [...] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] About the features.conf of it's transfer
HI This is my setup of the features.conf but it had not any reaction after I pushed the *2 while calling was acting ! Could you tell me the reason ? Or give my the method of the setting. Thanks! LARRY [general] parkext = 700 parkpos = 701-702 context = parkedcalls [featuremap] atxfer = *2 [applicationmap] set(DYNAMIC_FEATURES=tranf) tranf = *2,peer,waitexten(10|m) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 - 8.0.4SRS2 failing
Hi, My apologies for the OT. My googling came up empty and hopefully there are some members in the community that could give me a hint how to solve this issue: Cisco 7961 with SIP firmware 8.3.3. Needed to downgrade it to 8.0.4SRS2. The downgrade process started off good. The 7961 got it's IP address via DHCP, found it's SEPmac.cnf.xml file and started to upgrade the phone with the 8.0.4 firmware. All was well until it finally rebooted. Now it get's an IP from the DHCP server and says upgrading. Nothing else. It just seems to hang (monitored it for more than an hour). Anyone have an idea how I can fix this? Thanks and regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk realtime user deletion
Hi All, Would just like to know if anyone has encountered this: i a user is currently registered using SPA 941, i then tried deleting the user in the realtime db. then i tried to make a call from the SPA i can still make calls even though user has been deleted. i tried the same thing this time using an x-lite, i'm registered on x-lite, i deleted user in the db, x-lite cannot make calls, whcih should be the proper case. i tried same thing with zoiper, i got the same result as the x-lite. i have the rtcachefriends set to no, but why my SPA can still make calls when the xlite/zoiper cannot? TIA Regards Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transcoding
I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Avaya
I am told it is an IP Office 400 series. I have not been on site physically which does not help. Regards, Steve 2008/8/6 Tom Lynn [EMAIL PROTECTED]: Steve, what kind of Avaya system is this? They make several. On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote: Hi, Sorry this is so long, but I am reasonably desparate. I am having real fun with hooking an Avaya system to Asterisk using ISDN30. I have the ISDN signalling all sorted one way, and can pass calls from the real world (ie. the telco and asterisk) TO the avaya box, and it accepts that and sets up the call perfectly. The problem is that the Avaya box is signalling outbound calls using an odd method, which smacks of an analogue system with ISDN30 bolted on for a bit of a laugh. They send a q931 SETUP message. This contains the correct callerID, but only the first 1 to 4 of the dialled number's digits - The remainder of the number is I believe passed through using DTMF!!! From the look of it they intentionally do not send an IE 161 Sending Complete with the SETUP, so that the far end continues to listen for these DTMF tones, until it resolves to a legal number. My questions for some ISDN expert out there... Part 1) I need to receive the number in the SETUP, which I guess will be in ${EXTEN}, then I assume I can use Read() to collect DTMF digits, and check the dialplan to see if it is a locally terminated number. Once I am 100% sure it is not local, I can then dial the collected number through the Telco ISDN channel. Make sense? I think I can probably handle that. The problem is that I do not know whether I have received all digits from the Avaya at that point, which leads to... Part 2) Can I dial through Zaptel (via a Sangoma card if that makes a difference) without sending the IE 161 call complete? I thought that Dial(Zap/G1||D(${INITIAL})) might send the initial digits using DTMF, and then leave the channel open so that more DTMF could follow over the now bridged channel. In fact I get an immediate failure as if the far end thinks I have finished dialling. Can I assume that libpri does not currently support this method of dialling? If not, how might it be added? I can hack the code, I just need suggestions of where to look and how it might sanely be added :) Part 3) It is possible that the Avaya is not using DTMF at-all, and that it will send more bits of the called-party number using the D-Channel as you would expect, but Asterisk does not seem to be waiting for them. Can this be changed in Zaptel/Asterisk. Does anyone know the Avaya systems well enough to suggest how it might be working? Many many thanks for any feedback. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with iaxmodem!
Hello, I would like to configure hylafax(4.4.4) + iaxmodem(1.1.1). I use Asterisk and I work on Redhat. I installed the two hylafax and iaxmodem. My configuration of iaxmodem is: (in the file /etc/iaxmodem/ttyIAX0) device /dev/ttyIAX0 owner uucp:uucp mode 660 port 4570 #each line should have it's own port number! refresh 300 server 127.0.0.1 peername IAXmodem #this is the local extension number in FreePBX (create it!) secret 12345 #password for the extension cidname Fax1 cidnumber codec ulaw I added this two lines in /etc/inittab IA:2345:respawn:/usr/local/bin/iaxmodem ttyIAX0 mo:2345:respawn:/usr/sbin/faxgetty -D ttyIAX0 Then I tried to configure hylafax whith the command faxsetup I meet a problem when I want to add a modem with the command faxaddmodem but I can't, I have this response: Serial port that modem is connected to []? ttyIAX0 /dev/ttyIAX0 is not a terminal device. In fact in /dev I don't find ttyIAX0 I added the line /usr/sbin/faxgetty -D /dev/ttyIAX0 in the file /etc/rc.d//rc.local and I tried the command faxgetty -D /dev/ttyIAX0 but nothing!!! I tried the command /usr/local/bin/iaxmodem ttyIAX0 I have the following response: [EMAIL PROTECTED] /usr/local/bin/iaxmodem ttyIAX0 [2008-08-05 17:39:27] Modem started [2008-08-05 17:39:27] Setting device = '/dev/ttyIAX0' [2008-08-05 17:39:27] Setting owner = 'uucp:uucp' [2008-08-05 17:39:27] Setting mode = '660' [2008-08-05 17:39:27] Setting port = 4570 [2008-08-05 17:39:27] Setting refresh = 300 [2008-08-05 17:39:27] Setting server = '127.0.0.1' [2008-08-05 17:39:27] Setting peername = 'IAXmodem #this is the local extension number in FreePBX (create ' [2008-08-05 17:39:27] Setting secret = '12345 #password for the extension' [2008-08-05 17:39:27] Setting cidname = 'Fax1' [2008-08-05 17:39:27] Setting cidnumber = '' [2008-08-05 17:39:27] Setting codec = ulaw [2008-08-05 17:39:27] Opened pty, slave device: /dev/pts/17 [2008-08-05 17:39:27] Removed old /dev/ttyIAX0 [2008-08-05 17:39:27] Created /dev/ttyIAX0 symbolic link [2008-08-05 17:39:27] Registration failed. I don't unerstand why iaxmodem can't register . If someone has an idea, he is welcome!! Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding
I would make absolutely sure you've got your linux distro's version of libgsm installed. I can't really speak to the difference between those two versions of Asterisk without looking at a change-log, but I highly doubt a serious modification to the gsm code took place between sub- versions. Hope this helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 7:02 AM, Guilherme Loch Waltrick Góes wrote: I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with iaxmodem!
On Wed, Aug 6, 2008 at 4:05 PM, [EMAIL PROTECTED] wrote: Hello, I would like to configure hylafax(4.4.4) + iaxmodem(1.1.1). I use Asterisk and I work on Redhat. I installed the two hylafax and iaxmodem. My configuration of iaxmodem is: (in the file /etc/iaxmodem/ttyIAX0) device /dev/ttyIAX0 owner uucp:uucp mode 660 port 4570 #each line should have it's own port number! refresh 300 server 127.0.0.1 peername IAXmodem #this is the local extension number in FreePBX (create it!) secret 12345 #password for the extension cidname Fax1 cidnumber codec ulaw I added this two lines in /etc/inittab IA:2345:respawn:/usr/local/bin/iaxmodem ttyIAX0 mo:2345:respawn:/usr/sbin/faxgetty -D ttyIAX0 Then I tried to configure hylafax whith the command faxsetup I meet a problem when I want to add a modem with the command faxaddmodem but I can't, I have this response: Serial port that modem is connected to []? ttyIAX0 /dev/ttyIAX0 is not a terminal device. In fact in /dev I don't find ttyIAX0 I added the line /usr/sbin/faxgetty -D /dev/ttyIAX0 in the file /etc/rc.d//rc.local and I tried the command faxgetty -D /dev/ttyIAX0 but nothing!!! I tried the command /usr/local/bin/iaxmodem ttyIAX0 I have the following response: [EMAIL PROTECTED] /usr/local/bin/iaxmodem ttyIAX0 [2008-08-05 17:39:27] Modem started [2008-08-05 17:39:27] Setting device = '/dev/ttyIAX0' [2008-08-05 17:39:27] Setting owner = 'uucp:uucp' [2008-08-05 17:39:27] Setting mode = '660' [2008-08-05 17:39:27] Setting port = 4570 [2008-08-05 17:39:27] Setting refresh = 300 [2008-08-05 17:39:27] Setting server = '127.0.0.1' [2008-08-05 17:39:27] Setting peername = 'IAXmodem #this is the local extension number in FreePBX (create ' [2008-08-05 17:39:27] Setting secret = '12345 #password for the extension' [2008-08-05 17:39:27] Setting cidname = 'Fax1' [2008-08-05 17:39:27] Setting cidnumber = '' [2008-08-05 17:39:27] Setting codec = ulaw [2008-08-05 17:39:27] Opened pty, slave device: /dev/pts/17 [2008-08-05 17:39:27] Removed old /dev/ttyIAX0 [2008-08-05 17:39:27] Created /dev/ttyIAX0 symbolic link [2008-08-05 17:39:27] Registration failed. I don't unerstand why iaxmodem can't register . If someone has an idea, he is welcome!! Thank you What's your iax.conf? For me modem configuration looks like this: [iaxmodem5] type=friend host=dynamic secret=x context=fax permit=127.0.0.1 allow=all P.S. after editing inittab, you also have to execute # kill -HUP 1 So that init process re-reads configuration. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 - 8.0.4SRS2 failing
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Wednesday, August 06, 2008 7:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 - 8.0.4SRS2 failing Hi, My apologies for the OT. My googling came up empty and hopefully there are some members in the community that could give me a hint how to solve this issue: Cisco 7961 with SIP firmware 8.3.3. Needed to downgrade it to 8.0.4SRS2. The downgrade process started off good. The 7961 got it's IP address via DHCP, found it's SEPmac.cnf.xml file and started to upgrade the phone with the 8.0.4 firmware. All was well until it finally rebooted. Now it get's an IP from the DHCP server and says upgrading. Nothing else. It just seems to hang (monitored it for more than an hour). Anyone have an idea how I can fix this? Thanks and regards, Patrick Did you change your SEPXXX when you upgraded to 8.3.3? You may have to revert those changes. Check the debug log on the phones web interface to see if it's choking on a particular line in the cfg. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding
I'm using OpenSUSE 10.3, the funny thing is: if the softphone is using GSM the sounds is perfect, if I use Alaw as the softphone CODEC the sounds is pretty bad. The softphone is in the same LAN as the Asterisk server, so I don't think it's a bandwidth issue. Best Regards, On Wed, Aug 6, 2008 at 10:13 AM, Darren Sessions [EMAIL PROTECTED]wrote: I would make absolutely sure you've got your linux distro's version of libgsm installed. I can't really speak to the difference between those two versions of Asterisk without looking at a change-log, but I highly doubt a serious modification to the gsm code took place between sub-versions. Hope this helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 7:02 AM, Guilherme Loch Waltrick Góes wrote: I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding
I am a **BIG, BIG** fan of OpenSUSE. :) Use yast under 'Software Management' and do a search for 'gsm'. Make sure gsmlib and gsmlib-devel are *both* installed. Then scroll down and make sure that libgsm and libgsm-devel are *both* installed. After that, you'll have to recompile Asterisk. See if that does anything for you. - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 8:48 AM, Guilherme Loch Waltrick Góes wrote: I'm using OpenSUSE 10.3, the funny thing is: if the softphone is using GSM the sounds is perfect, if I use Alaw as the softphone CODEC the sounds is pretty bad. The softphone is in the same LAN as the Asterisk server, so I don't think it's a bandwidth issue. Best Regards, On Wed, Aug 6, 2008 at 10:13 AM, Darren Sessions [EMAIL PROTECTED] wrote: I would make absolutely sure you've got your linux distro's version of libgsm installed. I can't really speak to the difference between those two versions of Asterisk without looking at a change-log, but I highly doubt a serious modification to the gsm code took place between sub-versions. Hope this helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 7:02 AM, Guilherme Loch Waltrick Góes wrote: I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vars in Macros called by DIAL with option M
On Tue, Aug 05, 2008 at 02:13:15PM -0500, Tilghman Lesher wrote: above the original post is very confusing. Please stop doing this. The format of this post is in reverse, to demonstrate why posting a reply option is only in trunk. So no, it would not help him out. Yes, it works the same way, by using the U() option to Dial. However, this Question 2: prior to the origination of the slave channel. channel. No inheritance is possible, because the master channel originated channel, so any values set in the slave channel will not affect the master Apple. The variable is only set in the slave channel, not in the master Question 1: And the award for Best Illustration of a Point goes to... Tilghman Lesher! Mr Lesher has been nominated for this award 4 times; this is his first win. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding
Guilherme Loch Waltrick Góes wrote: I have a server with Asterisk 1.4.21.1 http://1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br One important difference between the servers may be the compiler used. We have heard reports that using GCC 4.2 or later with optimizations on causes choppy audio when using GSM. Solutions to this include either downgrading your compiler to GCC 4.1 or earlier, or selecting DONT_OPTIMIZE in menuselect under compiler options and then recompiling Asterisk. I also believe that you can set the optimization level for compilation to -O2 in Makefile.rules and have no choppy audio, but I cannot confirm this. Of course, if this server isn't running GCC 4.2, then you can ignore everything I've said so far :) Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding
On Wednesday 06 August 2008 08:13:09 Darren Sessions wrote: I would make absolutely sure you've got your linux distro's version of libgsm installed. I can't really speak to the difference between those two versions of Asterisk without looking at a change-log, but I highly doubt a serious modification to the gsm code took place between sub- versions. There was one slight change, which will only make a difference if you're using gcc 4.2 or above. The change was to fix a new optimization in gcc 4.2 that caused some inline assembly to be incorrectly built, which corrupted sound. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with iaxmodem!
My iax.conf looks like this: [iaxmodem] type=friend host=127.0.0.1 secret=x context=fax-out permit=127.0.0.1 disallow=all allow=ulaw after editing inittab I reload it by running: /sbin/init q I also reboot the system with shutdown -r now and I had the following message: init: Id mo respawning too fast: disabled for 5 minutes. I don't know what it signifies! Regards, Nadjia Boumediene, Legos [EMAIL PROTECTED] Work phone:+ 33172292995 -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Atis Lezdins Envoyé : mercredi 6 août 2008 15:46 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] problem with iaxmodem! On Wed, Aug 6, 2008 at 4:05 PM, [EMAIL PROTECTED] wrote: Hello, I would like to configure hylafax(4.4.4) + iaxmodem(1.1.1). I use Asterisk and I work on Redhat. I installed the two hylafax and iaxmodem. My configuration of iaxmodem is: (in the file /etc/iaxmodem/ttyIAX0) device /dev/ttyIAX0 owner uucp:uucp mode 660 port 4570 #each line should have it's own port number! refresh 300 server 127.0.0.1 peername IAXmodem #this is the local extension number in FreePBX (create it!) secret 12345 #password for the extension cidname Fax1 cidnumber codec ulaw I added this two lines in /etc/inittab IA:2345:respawn:/usr/local/bin/iaxmodem ttyIAX0 mo:2345:respawn:/usr/sbin/faxgetty -D ttyIAX0 Then I tried to configure hylafax whith the command faxsetup I meet a problem when I want to add a modem with the command faxaddmodem but I can't, I have this response: Serial port that modem is connected to []? ttyIAX0 /dev/ttyIAX0 is not a terminal device. In fact in /dev I don't find ttyIAX0 I added the line /usr/sbin/faxgetty -D /dev/ttyIAX0 in the file /etc/rc.d//rc.local and I tried the command faxgetty -D /dev/ttyIAX0 but nothing!!! I tried the command /usr/local/bin/iaxmodem ttyIAX0 I have the following response: [EMAIL PROTECTED] /usr/local/bin/iaxmodem ttyIAX0 [2008-08-05 17:39:27] Modem started [2008-08-05 17:39:27] Setting device = '/dev/ttyIAX0' [2008-08-05 17:39:27] Setting owner = 'uucp:uucp' [2008-08-05 17:39:27] Setting mode = '660' [2008-08-05 17:39:27] Setting port = 4570 [2008-08-05 17:39:27] Setting refresh = 300 [2008-08-05 17:39:27] Setting server = '127.0.0.1' [2008-08-05 17:39:27] Setting peername = 'IAXmodem #this is the local extension number in FreePBX (create ' [2008-08-05 17:39:27] Setting secret = '12345 #password for the extension' [2008-08-05 17:39:27] Setting cidname = 'Fax1' [2008-08-05 17:39:27] Setting cidnumber = '' [2008-08-05 17:39:27] Setting codec = ulaw [2008-08-05 17:39:27] Opened pty, slave device: /dev/pts/17 [2008-08-05 17:39:27] Removed old /dev/ttyIAX0 [2008-08-05 17:39:27] Created /dev/ttyIAX0 symbolic link [2008-08-05 17:39:27] Registration failed. I don't unerstand why iaxmodem can't register . If someone has an idea, he is welcome!! Thank you What's your iax.conf? For me modem configuration looks like this: [iaxmodem5] type=friend host=dynamic secret=x context=fax permit=127.0.0.1 allow=all P.S. after editing inittab, you also have to execute # kill -HUP 1 So that init process re-reads configuration. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Action on login
Hi, is there meanwhile the possibility for some actions besides dialling in *? Namely, I would like that if a remote IAX or SIP user logs in AND there are new messages, they automatically get a call and be connected to the voicemail. The only method I know by now is make a context in the dialplan, checking if the user has logged in and then initiate the call. And of course firing a callfile to every x minutes to that context for each remote user. That does not scale very well. It would be much nicer to have some kind of login / logout action parameter in sip.conf or so. --Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding
I have used virtually all versions of Asterisk 1.0+ (literally, either in production or testing) with OpenSUSE 10+ and 11 on AMD and Intel and haven't had any issues with gcc optimizations with regards to audio sounding choppy. This scenario for me has always been the gsm libs. _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 9:16 AM, Mark Michelson wrote: Guilherme Loch Waltrick Góes wrote: I have a server with Asterisk 1.4.21.1 http://1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br One important difference between the servers may be the compiler used. We have heard reports that using GCC 4.2 or later with optimizations on causes choppy audio when using GSM. Solutions to this include either downgrading your compiler to GCC 4.1 or earlier, or selecting DONT_OPTIMIZE in menuselect under compiler options and then recompiling Asterisk. I also believe that you can set the optimization level for compilation to -O2 in Makefile.rules and have no choppy audio, but I cannot confirm this. Of course, if this server isn't running GCC 4.2, then you can ignore everything I've said so far :) Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with iaxmodem!
Hi, Am Mittwoch, den 06.08.2008, 17:24 +0200 schrieb Nadjia Boumédiène: My iax.conf looks like this: [iaxmodem] type=friend host=127.0.0.1 secret=x context=fax-out permit=127.0.0.1 disallow=all allow=ulaw after editing inittab I reload it by running: /sbin/init q I also reboot the system with shutdown -r now and I had the following message: init: Id mo respawning too fast: disabled for 5 minutes. this message indicates, that the service (identified through id mo) died short after it's start. So the init-process starts it again and again. Cause this happens to fast, it disables the restarting of the process. Having a look at Your inittab in Your first post, I would suggest to remove the -D switch from the line with faxgetty. This command line switch instructs the faxgetty to detach from terminal. In this case the init process looses contact to the process and tries to restart it. HTH, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 14
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will be on vacation until Tuesday 19th of August with limited access to voice and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on my return. Dimitri Osler ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 - 8.0.4SRS2 failing
Hi Matt, Thank you for your suggestion. Comment inline. Matt Gibson wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Wednesday, August 06, 2008 7:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 - 8.0.4SRS2 failing Hi, My apologies for the OT. My googling came up empty and hopefully there are some members in the community that could give me a hint how to solve this issue: Cisco 7961 with SIP firmware 8.3.3. Needed to downgrade it to 8.0.4SRS2. The downgrade process started off good. The 7961 got it's IP address via DHCP, found it's SEPmac.cnf.xml file and started to upgrade the phone with the 8.0.4 firmware. All was well until it finally rebooted. Now it get's an IP from the DHCP server and says upgrading. Nothing else. It just seems to hang (monitored it for more than an hour). Anyone have an idea how I can fix this? Thanks and regards, Patrick Did you change your SEPXXX when you upgraded to 8.3.3? You may have to revert those changes. Check the debug log on the phones web interface to see if it's choking on a particular line in the cfg. Just to make sure, it's a downgrade from 8.3.3 to 8.0.4. The only line changed in the SEPxxx file is: from loadInformationSIP41.8-3-3S/loadInformation to loadInformationSIP41.8-0-4SRS2/loadInformation Nothing else changed. When I power the phone down and up again, the only thing it does is going into the upgrade screen, getting an IP from the DHCP server and then say Upgrading. In the tftpserver logs I can see that it does not even pick up the SEPxxx file. I haven't checked the phone's web interface. Frankly I wasn't aware that it was active during the upgrade process. I did try telnet and ssh but both were unresponsive. Will try the web interface when I'm near the phone again. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] in-call start monitoring
I suppose, too. So see below. I also verified that the dial command is using Ww (which I had to fudge), but still, no monitoring. Anything else I can check? pig*CLI feature show Builtin Feature Default Current --- --- --- Pickup *8 *8 Blind Transfer # # Attended Transfer One Touch Monitor *1 Disconnect Call * ** Park Call Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 70 Parking context : parkedcalls Parked call extensions: 71-79 From: Paul Hales [EMAIL PROTECTED] I suppose the bit to check is the features ('show features') and then try to record a call (*1) and see what the terminal says... Bill Michaelson wrote: My client needs call recording features and would like to initiate the process in-call (typically *1). I'm installing Asterisk 1.4.x and FreePBX 2.4+. I'm using Polycom phones. I can't make it work. Would somebody please give a checklist of items for me to compare my list against - in the hope I've overlooked something? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Avaya
2008/8/5 Steve Davies [EMAIL PROTECTED]: Hi, Sorry this is so long, but I am reasonably desparate. I am having real fun with hooking an Avaya system to Asterisk using ISDN30. I have the ISDN signalling all sorted one way, and can pass calls from the real world (ie. the telco and asterisk) TO the avaya box, and it accepts that and sets up the call perfectly. The problem is that the Avaya box is signalling outbound calls using an odd method, which smacks of an analogue system with ISDN30 bolted on for a bit of a laugh. [...] Okay, I think I am progressing in terms of my understanding. Firstly, I had missed out overlapdial=yes from the inbound PRI_NET channel from the Avaya. I have not been able to check, but that should allow Asterisk to collect the remaining digits until it finds a match in the dialplan. This begs the following questions: If an inbound overlapdial uniquely matches: exten = _X.,1,NoOp() Then I assume it will match after any 6 digits have been received, and drop into the dialplan. Given that this is a Zap channel, how do I receive any subsequent digits if they are dialled? Are they converted to inband DTMF??? I cannot find any useful documentation on what overlapdial=yes really does - Pointers welcome. Also, what if the overlap dialled number will never be unique, so I need to trap both: exten = _X.,1,NoOp() and exten = 01234567890,1,NoOp() Will overlapdial ever start executing one of those 2 patterns if I dial 01234567890 and nothing else? I appreciate that I could probably do the following instead - perhaps I've answered my own question? exten = 01234567890,1,NoOp() exten = _!.,1,Goto(passthru,${EXTEN},1) Then, importantly, how do I overlap dial outbound using Zaptel? The Dial() command is designed to send a number and wait for a response. There is no opportunity for further input AFAIK. Does enabling overlapdial=yes mean that I can Dial() and it will not assume the number is complete? Perhaps Asterisk simply cannot do this? It is a pretty horrible requirement! Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Regarding fmtp parameters.
Hello All, I'am doing a video call between two Video Phones, and i see that Asterisk is stripping the fmtp parameters for the h263 video line in SDP. For example a line similar to the below is stripped, a=fmtp:xx CIF=4;QCIF=2;F=1;K=1 Asterisk is configured NOT to be present in the Media path (My version : Asterisk 1.4.19.1 ). I have the following enabled in my sip.conf. canreinvite=yes directrtpsetup=yes From what i have read on the internet, i feel fmtp parameters are not supported by Asterisk for Video. I also find that video_caps branch has a fix for this problem, please can someone share more information about this and where i can find it ? I do not want those fmtp lines to be stripped. Suggestions to change the Asterisk config files, to achieve this are also welcome. Thank you. Best regards, Simith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 15
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will be on vacation until Tuesday 19th of August with limited access to voice and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on my return. Dimitri Osler ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding
On Wed, Aug 06, 2008 at 10:15:00AM -0500, Tilghman Lesher wrote: On Wednesday 06 August 2008 08:13:09 Darren Sessions wrote: I would make absolutely sure you've got your linux distro's version of libgsm installed. I can't really speak to the difference between those two versions of Asterisk without looking at a change-log, but I highly doubt a serious modification to the gsm code took place between sub- versions. There was one slight change, which will only make a difference if you're using gcc 4.2 or above. The change was to fix a new optimization in gcc 4.2 that caused some inline assembly to be incorrectly built, which corrupted sound. Also note that the issue was triggered by uisng -O3. I know that at least most Debian packages are built with -O2 . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Max amount of concurrent calls on and iax trunk
hi, wanted to ask if anybody has experienced setting up two asterisk 1.2 boxes connected via iax trunk. have u guys ever stress tested the trunks i.e how many concurrent calls can a trunk handle and whether codec has any effect on it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange beep during calls
Hi all, Our users are complaining about beeps that happen in the middle of some calls. They are similar to the sound heard you are in a call and press any button in your phone. Please find bellow some examples of these beeps(the recordings are in Portuguese, but the beeps are easy to identify): http://www.katizak.locaweb.com.br/asterisk/beep.mp3 http://www.katizak.locaweb.com.br/asterisk/beep.mp3 http://www.katizak.locaweb.com.br/asterisk/beep2.mp3 http://www.katizak.locaweb.com.br/asterisk/beep3.mp3 http://www.katizak.locaweb.com.br/asterisk/beep4.mp3 We are sure that our users are not pressing any button in the softphones during the conversations. Do you guys are able to identify where these beeps are coming from? Maybe an * functionality that we need to turn off... We are using Asterisk 1.4.21.2. Thanks. Felippe Silvestre ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange beep during calls
maybe you are using the L option in Dial app to limit the conversation time. Check those channel variables (just a wild guess) *LIMIT_PLAYAUDIO_CALLER **LIMIT_PLAYAUDIO_CALLEE **LIMIT_TIMEOUT_FILE **LIMIT_CONNECT_FILE **LIMIT_WARNING_FILE * Felippe Silvestre wrote: Hi all, Our users are complaining about beeps that happen in the middle of some calls. They are similar to the sound heard you are in a call and press any button in your phone. Please find bellow some examples of these beeps(the recordings are in Portuguese, but the beeps are easy to identify): http://www.katizak.locaweb.com.br/asterisk/beep.mp3 http://www.katizak.locaweb.com.br/asterisk/beep2.mp3 http://www.katizak.locaweb.com.br/asterisk/beep3.mp3 http://www.katizak.locaweb.com.br/asterisk/beep4.mp3 We are sure that our users are not pressing any button in the softphones during the conversations. Do you guys are able to identify where these beeps are coming from? Maybe an * functionality that we need to turn off... We are using Asterisk 1.4.21.2. Thanks. Felippe Silvestre ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max amount of concurrent calls on and iax trunk
Rosli Sukri wrote: hi, wanted to ask if anybody has experienced setting up two asterisk 1.2 boxes connected via iax trunk. have u guys ever stress tested the trunks i.e how many concurrent calls can a trunk handle and whether codec has any effect on it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users What are the hardware specs of the boxes, and what is the speed of the connection between them? -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk realtime user deletion
Hi All, Would just like to know if anyone has encountered this: i a user is currently registered using SPA 941, i then tried deleting the user in the realtime db. then i tried to make a call from the SPA i can still make calls even though user has been deleted. i tried the same thing this time using an x-lite, i'm registered on x-lite, i deleted user in the db, x-lite cannot make calls, whcih should be the proper case. i tried same thing with zoiper, i got the same result as the x-lite. i have the rtcachefriends set to no, but why my SPA can still make calls when the xlite/zoiper cannot? TIA Regards Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium B410P: problematic Bri connection between * and a legacy Philips PBX
Hi all, my goal is to connect my trixbox server (CentOS 5.2 - kernel 2.6.18-53.1.4.el5 - * 1.4.20-1) with a legacy Philips PBX with 4 bri links provided from Digium B410P. For this reason I set all the 4 ports of Digium's card in NT mode (Philips can not do this). Then i opportunely edited /etc/misdn-init.conf and /etc/asterisk/misdn.conf. In fact, when I run the command misdn shows stacks in * CLI, I can see all ports in NT (PTP) mode. Once i connect the wires from Philips PBX to the Digium's card, L2 and L1 go immediately up and i can do any calls from * to Philips and viceversa...only for 10 minutes! Then from Philips console I can see that B-channels of every port become not usable. From that moment any calls form Philips to * fail but: 1) I can still place calls from * to Philips 2) misdn show stacks output does not underline any problem (all 4 ports appear up with L2/L1). In your opinion, how can i fix this? why this problem after 10 minutes? Thank you - Daniele p.s.: I already attempt to change some options in /etc/asterisk/misdn.conf file. I tried to put incoming_early_audio=yes (explanation: Rarely used. If turned on, sends Tone Indications on TE Port for Incoming isdn channel. Normally the telcos send that informaton. By default is 'no'). It didn't work... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk realtime user deletion
On Wednesday 06 August 2008 15:07:03 Nhadie wrote: Would just like to know if anyone has encountered this: You sent the exact same email this morning at 7:47 a.m. If nobody has responded, it's because nobody has ever seen that before. Duplicating the message tells everybody on the list that you have low regard for their time and bandwidth. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] intercom/paging with grandstream gxp2000
Guys I have been reading for days on how to get this to work with asterisk and for some reason every time I call the call goes to intercom. I know I must be doing something wrong with the way I am adding the steps to my call; I am not familiar with variables and flags. Here is my configuration: Digium Asterisk AA50 with Granstream GXP2000 using the latest firmware. Extensions.conf: exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family) exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3) exten=s,2,SIPAddHeader(Call-Info: answer-after=0) exten=s,3,Dial(${ARG2},20) exten=s,4,Goto(s-${DIALSTATUS},1) exten=s-NOANSWER,1,Voicemail(${ARG1},u) exten=s-NOANSWER,2,Goto(default,s,1) exten=s-BUSY,1,Voicemail(${ARG1},b) exten=s-BUSY,2,Goto(default,s,1) exten=_s-.,1,Goto(s-NOANSWER,1) exten=a,1,VoicemailMain(${ARG1}) GXP2000 configuration: Under Account1 I checked options: Allow Auto Answer by Call-Info: No Yes Turn off speaker on remote disconnect: No Yes Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intercom/paging with grandstream gxp2000
I am sorry, this is the actual extensions.conf: exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family) exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3) exten=s,3,SIPAddHeader(Call-Info: answer-after=0) exten=s,4,Dial(${ARG2},20) exten=s,5,Goto(s-${DIALSTATUS},1) exten=s-NOANSWER,1,Voicemail(${ARG1},u) exten=s-NOANSWER,2,Goto(default,s,1) exten=s-BUSY,1,Voicemail(${ARG1},b) exten=s-BUSY,2,Goto(default,s,1) exten=_s-.,1,Goto(s-NOANSWER,1) exten=a,1,VoicemailMain(${ARG1}) As you can see here Goto and SIPAddHeader are 2 and 3. In the prior email I had both lines under 2. Fidel Garcia From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fidel Garcia Sent: Wednesday, August 06, 2008 5:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] intercom/paging with grandstream gxp2000 Guys I have been reading for days on how to get this to work with asterisk and for some reason every time I call the call goes to intercom. I know I must be doing something wrong with the way I am adding the steps to my call; I am not familiar with variables and flags. Here is my configuration: Digium Asterisk AA50 with Granstream GXP2000 using the latest firmware. Extensions.conf: exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family) exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3) exten=s,2,SIPAddHeader(Call-Info: answer-after=0) exten=s,3,Dial(${ARG2},20) exten=s,4,Goto(s-${DIALSTATUS},1) exten=s-NOANSWER,1,Voicemail(${ARG1},u) exten=s-NOANSWER,2,Goto(default,s,1) exten=s-BUSY,1,Voicemail(${ARG1},b) exten=s-BUSY,2,Goto(default,s,1) exten=_s-.,1,Goto(s-NOANSWER,1) exten=a,1,VoicemailMain(${ARG1}) GXP2000 configuration: Under Account1 I checked options: Allow Auto Answer by Call-Info: No Yes Turn off speaker on remote disconnect: No Yes Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.5.12/1595 - Release Date: 8/6/2008 8:23 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange beep during calls
Felippe Silvestre wrote: Hi all, Our users are complaining about beeps that happen in the middle of some calls. They are similar to the sound heard you are in a call and press any button in your phone. Please find bellow some examples of these beeps(the recordings are in Portuguese, but the beeps are easy to identify): http://www.katizak.locaweb.com.br/asterisk/beep.mp3 http://www.katizak.locaweb.com.br/asterisk/beep2.mp3 http://www.katizak.locaweb.com.br/asterisk/beep3.mp3 http://www.katizak.locaweb.com.br/asterisk/beep4.mp3 We are sure that our users are not pressing any button in the softphones during the conversations. Do you guys are able to identify where these beeps are coming from? Maybe an * functionality that we need to turn off... We are using Asterisk 1.4.21.2. Thanks. There was a short discussion on the OSLEC mailing list very recently about something that sounds (forgive the pun) similar. (Sorry I can't add anything else but I deleted them.) I can't recall what the suggestions were but I think someone mentioned possible hardware faults on an analogue line card... Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange beep during calls
I've also seen systems where the IRQ between the card and another heavily loaded device (disk controller) are shared causing clicks, beeps, and pops to be present in the audio stream. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Alan Lord [EMAIL PROTECTED] wrote: Felippe Silvestre wrote: Hi all, Our users are complaining about beeps that happen in the middle of some calls. They are similar to the sound heard you are in a call and press any button in your phone. Please find bellow some examples of these beeps(the recordings are in Portuguese, but the beeps are easy to identify): http://www.katizak.locaweb.com.br/asterisk/beep.mp3 http://www.katizak.locaweb.com.br/asterisk/beep2.mp3 http://www.katizak.locaweb.com.br/asterisk/beep3.mp3 http://www.katizak.locaweb.com.br/asterisk/beep4.mp3 We are sure that our users are not pressing any button in the softphones during the conversations. Do you guys are able to identify where these beeps are coming from? Maybe an * functionality that we need to turn off... We are using Asterisk 1.4.21.2. Thanks. There was a short discussion on the OSLEC mailing list very recently about something that sounds (forgive the pun) similar. (Sorry I can't add anything else but I deleted them.) I can't recall what the suggestions were but I think someone mentioned possible hardware faults on an analogue line card... Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max amount of concurrent calls on and iax trunk
I have two Asterisk 1.4 boxes connected via IAX over a VPN tunnel on a 10Mbit link. We never did any stress testing as it's a temporary arrangement, but we've never had any call quality issues or run up against concurrent call limitations. I'm mostly routing internal extensions over the trunk, and in the case of two floating users I have their extensions at each office ring when their DID is called. One server is an older Pentium 4 1.7 GHz with 1GB Ram, and the other is a Dual Xeon 2.33 GHz with 4GB Ram. As for codec, I'm disallowing all except ulaw and gsm, with ulaw the priority codec for hardphones (Polycom) and gsm the priority for softphones (X-Lite, Zoiper). I would expect the limitation you're going to run up against is not Asterisk, but the bandwidth between your two systems. On 6 Aug, 2008, at 10:40 AM, Rosli Sukri wrote: hi, wanted to ask if anybody has experienced setting up two asterisk 1.2 boxes connected via iax trunk. have u guys ever stress tested the trunks i.e how many concurrent calls can a trunk handle and whether codec has any effect on it. ATT1.c ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trying to understand Messages from chan_zap.c
Hi friends, Where can I get some information to understand messages like the following ones? *NOTICE[6455] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1* *NOTICE[6455] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1* * ERROR[6455] chan_zap.c: !! Got S-frame while link down* *ERROR[6458] chan_zap.c: !! Got reject for frame 27, retransmitting frame 27 now, updating n_r!* *ERROR[6458] chan_zap.c: !! Got a UA, but i'm in state 7* *ERROR[6455] chan_zap.c: ACK received for '1' outside of window of '0' to '0', restarting* * ERROR[6455] chan_zap.c: !! Not good - head of queue has not been transmitted yet* *ERROR[6457] chan_zap.c: !! Got reject for frame 52, but we only have others!* Yesterday I have had several voice problems with my calls but my telephone provider says he had no problems then but I don't think so. Today everything was fine and I didn't do any change. I'm using Asterisk 1.4.21.1 (very good with queue delivery), Zaptel 1.4.11 and Digium T412P card. Thanks in advance! Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Capture digits, set as variable..., use for caller id?
We've searched but thus far have not successfully found a solution for this… We're looking for a way to set a variable using get digits for a DISA application. Sometimes we're away from the office and get a voicemail that I need to respond to quickly and would prefer for the caller to be presented with the caller id of the office, or perhaps home…. I would like to set up DISA so that we can dial into the switch, enter a password, provide the outgoing caller ID that we want to present, enter the number I want to dial, and PRESTO.. make a call… Any ideas? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk realtime user deletion
i apologize, coz i had some experiences that my mail did not go thru. again i apologize. Tilghman Lesher wrote: On Wednesday 06 August 2008 15:07:03 Nhadie wrote: Would just like to know if anyone has encountered this: You sent the exact same email this morning at 7:47 a.m. If nobody has responded, it's because nobody has ever seen that before. Duplicating the message tells everybody on the list that you have low regard for their time and bandwidth. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Randulo: An open suggestion for the VOIP users Conference
Randy: Kudos to you for running the outstanding VOIP user's conference. I have an idea to toss into this public forum. I'm hoping that you and others will consider it give some feedback. The idea would be to begin each show with comments corrections from the previous week's show. Sometimes when I listen to the previous week's archive, I find that there is misinformation that could be corrected, or even a 'dangling' question that nobody on the conference could answer. Someone participating in the FOLLOWING week may be more inclined to comment, correct, and even expand on such things if there were an official place for it. It may also make the first part of the show more dense with specific useful information rather than being more free-form. Doubtless someone going through the archives would look forward to the beginning of the NEXT archive which would start off with dense ( corrected) key points of the previous call. Example: Last week there was talk about Polycom's HDVoice technology, and the term was being used interchangeably with G.722. In fact there are important distinctions, but someone listening might presume that the information was correct and leave short-changed. There are other examples even from last week, one involving someone's claim that there's not a way to pick up a phone and directly interface with a voice recognition directory application without needing to press some digits first. As it turns out, it's easy if you know the trick. Id' be happy to put my money where my mouth is and kick off this Friday's show with these examples any others I'm not remembering at this moment if you think it would be well received. Perhaps others will do the same. What do you think? Thanks! -Karl Fife If you want to discuss this off-list, you can email me at [EMAIL PROTECTED] p.s. As it turns out, HDVoice CAN use G.722, but it can also be overlain onto other codec's such as use G.722.1 and even G.711µ [sic]. That's right, you can have an HDVvoice call over the PSTN using G.711, using a special companding overlay on top G.711. As I understand it, the two HDVoice compliant endpoints (Polycom, Cisco others that license the technology) have an in-band (but inaudible) handshake, and then begin applying the proprietary companding overlay which extends the dynamic range of the audio. It sounds great even though the underlying codec is not a wideband codec. Certainly the sound is not as good as HDVoice over a modern adaptive-transform codec like G.722 (1987) or even better over G.722.1 (1999), but it's definitely a big improvement over the Toll-Quality (Read: AM-Radio-Through-A-Pillow) that we're all used to, and it is not dependent upon having a pure-IP connection involving ENUM, DUNDI, or other non e.164 namespaces such as SIP URI's, ITAD Subscriber Numbers etc. In my opinion HDVoice is it's a brilliant transition technology. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users