[asterisk-users] Dahdi problem with dahdi_genconf
Hi, It's the first time i try to configure an ISDN card with dahdi, so my experience is very poor (be kind ;)) My problem is with dahdi_genconf, when i start it it says: /usr/sbin/dahdi_span_assignments: Missing '/sys/bus/dahdi_devices/devices' (DAHDI driver unloaded?) Command failed (status=256): 'dahdi_span_assignments dumpconfig /etc/dahdi/assigned-spans.conf' at /usr/local/share/perl/5.18.2/Dahdi/Config/Gen/Assignedspans.pm line 40. Here is my dahdi_scan: [1] active=yes alarms=UNCONFIGURED description=HFC-S PCI A ISDN card 0 [TE] name=ZTHFC1 manufacturer=Cologne Chips devicetype=HFC-S PCI-A ISDN location=PCI Bus 01 Slot 10 basechan=1 totchans=3 irq=18 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI framing_opts=CCS coding= framing=CAS dahdi_hardware pci::01:09.0 zaphfc+ 1397:2bd0 HFC-S ISDN BRI card lsmod zaphfc 22268 0 dahdi 204045 1 zaphfc DAHDI Version: 2.5.0.1 Echo Canceller: HWEC Obviously in asterisk, there is no channel configured: *CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState Description pseudo defaultdefault My big problem is i can't test the ISDN line with a phone (i don't have an ISDN phone), so my dubt now is the NT1 is not working correctly... Any suggestion about this ? Thank you in advance. Cordially, Claudio. -- _ -- 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] Dahdi problem with dahdi_genconf
Il 29/09/2014 15:57, Tzafrir Cohen ha scritto: On Mon, Sep 29, 2014 at 03:52:25PM +0200, Claudio ML wrote: Hi, It's the first time i try to configure an ISDN card with dahdi, so my experience is very poor (be kind ;)) My problem is with dahdi_genconf, when i start it it says: /usr/sbin/dahdi_span_assignments: Missing '/sys/bus/dahdi_devices/devices' (DAHDI driver unloaded?) What version of the DAHDI drivers is loaded? cat /sys/module/dahdi/version cat /sys/module/dahdi/version 2.5.0.1 It seems the right one. -- _ -- 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] Ubuntu 14.04 LTS Asterisk and ISDN Cologne Chip
Hi to all, I am searching to make work an Asterisk, with an ISDN card with Cologne Chipset. Here is the lspci: 01:09.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Cologne Chip Designs GmbH ISDN Board Flags: bus master, medium devsel, latency 16, IRQ 5 I/O ports at c400 [size=8] Memory at fdefd000 (32-bit, non-prefetchable) [size=256] Capabilities: [40] Power Management version 1 I think i need the zaphfc module to make it work, but it is not included into the dahdi package, and i cant find it. The command dahdi_hardware says this: dahdi_hardware pci::01:09.0 zaphfc- 1397:2bd0 HFC-S ISDN BRI card But, how i can install the zaphfc module? Thanks in advance to all. Claudio. -- _ -- 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] better timing source for an asterisk gateway
Hi, I have to make an asterisk gateway in front of several other asterisk. This gateway will essentialy be used for outbound call. This gateway will be connected to other asterisk by IAX trunk, outbound call will use SIP trunk (voip provider or patton isdn). I have a TE220BF available than i can use for dahdi timing source. Is a good idea, or this will give me zero benefit for timerfd timing source (will host this gateway on debian squeeze or centos 6.2) ? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi and digium debian package
Hi I'm trying to install dahdi. I just need the dahdi timer for conference. I currently using digium debian package for asterisk 1.8.8.1. When i install asterisk-dahdi , i've got several dependencies which came for official debian repository (including the dahdi package) and are outdated. Is it normal than dahdi is not include into digium packages ? Do i have to compil it before install asterisk-dahdi ? Thanks for your help. -- _ -- 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] smsq, Zaptel in UK
Hi all, I've been trying to get SMS operational on my Asterisk box, which has a TDM400P card with a pair of FXO interfaces configured (ZAP/1 ZAP/2). I've not had luck with either of my lines, after issuing the command smsq --motx-channel=ZAP/1/1709400X 0 register. I see the following output in my Asterisk console: -- Attempting call on ZAP/1/17094009 for application SMS(0) (Retry 1) -- Hungup 'Zap/1-1' [Dec 26 18:17:07] NOTICE[526]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason (1) Hangup It keeps retrying with the same message as above until giving up. It doesn't seem to make any difference whether I specify ZAP/1 or ZAP/2. When I look in /var/spool/asterisk/outgoing, I see a file with the following content: Channel: ZAP/1/17094009 Callerid: SMS 0 Application: SMS Data: 0 MaxRetries: 10 RetryTime: 1 WaitTime: 10 In /var/spool/asterisk/sms/motx I see a corresponding file with the following contents: da=0 ud=register I'm probably missing something really obvious, but I've not found anything via Google that suggests what I'm doing wrong. I'm running Asterisk 1.4.14 Zaptel 1.4.6 on Ubuntu 7.10. Any help would be appreciated. Cheers, Chris -- Chris Notley [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem, to register an ata on an asterisk
Hello, I hope, somone can help me. When I try to register an ata (sipura 2002 for example), it is successfull, when a device is installed since a few weeks on the asterisk. It isn´t successfull, when it is a completly new device, I added to the asterisk. Have someone any idea? Many thanks! Hubertus Niewel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXy Hung, Power-cycle Required
Has anyone had good success with the IAXy? I've tried everything including PAT on the IAX2 port to the IAXy device to no avail (using the alternate server parameter). I guess a call to Digium is in order! Regards, --- Gavin Adams Promisant (USA) Inc. O: 770-913-3727 F: 770-913-3726 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryan Field-Elliot Sent: Tuesday, February 01, 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXy Hung, Power-cycle Required On Wed, 2005-01-26 at 15:59 -0500, Paul Dugas wrote: I've got a single IAXy installed in a little office nearby and got a call from someone on site a finew mintues ago. Apparently they couldn't make a call on that extension. They'd pick up the phone and get nothing; no dial-tone. Has snyone else had trouble with these things sticking like this? Paul Yes - we are having the exact same problem with a portion of our IAXys in the field. In all cases the IAXys are behind simple SOHO firewalls like the Linksys. After an idle period - perhaps 1-3 days - they just stop working, in both directions, but a simple power cycle restores functionality. We have an open support incident with Digium but have not yet heard back. FWIW we have stopped selling deploying the IAXys until we have a resolution to the problem. Bryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXy Configuration for Alternate Server
Hi, I've provisioned an IAXy adapter on a network segment local to my asterisk server. Provisioning is fine, as is the registration and use of said device. Since the local address is private address space, I setup the public IP address of my Asterisk server as the alternate. When taking it to another location (NAT again behind a WRT54G), it doesn't seem to work. Here's my current iaxprov.conf file that I downloaded to said IAXy: [default] server=172.16.200.3 altserver=65.x.x.x codec=ulaw flags=register,heartbeat tos=lowdelay [iaxy1] user=iaxy1 pass=xx template=default And the IAXY is online: coke*CLI iax2 show peers Name/UsernameHost Mask Port Status iaxy1172.16.200.38 (D) 255.255.255.255 4569 OK (3 ms) voicepulse 66.234.228.132 (S) 255.255.255.255 4569 OK (34 ms) Any pointers on how the altserver logic works? Optimal solution is that when local is registers against the private address space of my server and when on the road hits the alternate, or public IP address of my Asterisk server. Oh, apologies for the earlier test messages, my system admin was testing send on behalf. Grrr. Regards, --- Gavin Adams Promisant (USA) Inc. O: 770-913-3727 F: 770-913-3726 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] test
testing Will Stowe Systems Administrator Promisant (USA) Inc. email:[EMAIL PROTECTED] Office: (770) 913-3723 Mobile: (404) 993-0526 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk
Will Stowe Systems Administrator Promisant (USA) Inc. email:[EMAIL PROTECTED] Office: (770) 913-3723 Mobile: (404) 993-0526 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxComm version 1.0 released
Or you can try getting it at: http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-1.0rc1.zip http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-lin-1.0rc1.tar :) http://iaxclient.sourceforge.net/iaxcomm-win-1.0rc1.zip http://iaxclient.sourceforge.net/iaxcomm-lin-1.0rc1.tar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceConduits - Notice
Hello, This is David Deutsch, and Im the owner of VoiceConduits. There seems to be some confusion related to our company, regarding the past few posts. VoiceConduits is currently NOT open for public business, we have never to date advertised or attempted to attract business. It appears that a few people heard about our company via a mention in a SineApps article and found our beta system that is under development. We apologize that a few people managed to sign up via this interface, and we will happily refund anyone who did so immediately, additionally we will supply them with free credit to be used once we are in fact live. It was certainly never our intention to defraud individuals of the asterisk or voip community, our understanding is that only 5 people have managed to signup thru this automated system, and we will be contacting each of them individually to insure they are refunded and happy with the resolution. Thank you, David Deutsch, President Tris Telecommunications, LLC (800) 547-4057 x1001 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] After RC1 upgrade, temporary loss of voice
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] I just upgraded to RC1 from a two-three month old CVS , and noticed that during IAX2 calls to my service provider there are periods (usually less than 10 seconds long, minutes apart) during which the caller can not hear me, but I can hear the caller fine. Inter-office calls (SIP-to-SIP) does not appear to have this issue. Has any other users experienced this? Yup, We're experiencing it here on at 1.0-RC1 upgrade. SIP to SIP calls are rock solid uLAW/aLAW, but IAX2 to NuFone (and assumably other IAX2-IAX2 connections) are giving the breakups. Not QoS related as our uLAW/aLAW connections transit the same routes as the IAX2 traffic (using GSM or iLBC). Will try to find the commands to update to the latest CVS source for 1.0-RC1. Are the appropriate CVS commands on the Wiki? Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue feature
What I'm looking for is the ability to determine whether or not a queue has any queue handlers (active agents), and if it does not, bypass sending the caller to the queue and pass them on to a message or IVR system. -Chris http://bugs.digium.com/bug_view_page.php?bug_id=214 This is not exactly what you are looking for, but you can set a queue timeout. Post a request to that bug for the feature. Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug 789 - Announce/Music on Hold
Hi. I have posted a fix for announce so that it does not stop the music on hold until after playing the announcement file. If you can, please test it out for me. Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Termination in Phoenix
Hi. I'm looking for simple residential Asterisk use. I will need a local DID number in the Phoenix, AZ area which includes the area codes 480, 602, 623. Can anyone provide termination in those area codes with SIP or IAX? I suppose H.323 is also ok, but I haven't messed with it. Is a voicemail option available if I don't answer or not registered? Thanks, Kevin Bockman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec matching weirdness
I am experiencing a problem that from list archive it appears others are running into. When I dial from Cisco 7960 via the * to Free World Dialup destinations that supports G.729 the call fails. The major error from the debug log is Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format: Unable to find a path from G729A to ULAW Jan 15 00:11:14 NOTICE[22545]: channel.c:1451 ast_set_write_format: Unable to find a path from ULAW to G729A Me too? I've been wondering the same thing. I asked before and didn't really get anywhere either. Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re Grandstream 1.0.4.38
I got 1.0.4.38 from SIPphone's server at 130.94.123.253 but last time I tried it was offering 1.0.4.35. For me 1.0.4.38 cleared all my problems but from SIPphone's email it had hosed some phones, such that they were talking about replacement units. I'm still on fscking 1.0.3.81, and I can't get anything from the server above... can someone tell me how I can get a newer version? that's the latest on gs's pages as well, and it's really really buggy roy Another nice Asterisk user posted 1.0.4.39 on their website. I really like it. I haven't had much time to test it, but it is good so far. http://www.supercomputo.com/b13p4.39.zip Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LCR / Trollphone Rate Engine
Hi. Thank you to Olle, the Wiki, and Trollphone. I found the Trollphone Rate Engine listed on the Wiki at: http://www.voip-info.org/wiki-Asterisk+addon+rate-engine I'm not familiar with LCR yet, but this is something that I need to do. I have it all installed but am not familiar with the terminology. Can someone help me maybe figure out what some of these fields are for? mysql describe egress; +-+--+--+-+-++ | Field | Type | Null | Key | Default | Extra | +-+--+--+-+-++ | route_id| int(10) unsigned | | PRI | NULL| auto_increment | | provider_id | int(10) unsigned | | MUL | 0 || | technology | varchar(16) | | | || (I assume this should be SIP, IAX, etc) | peer| varchar(32) | | | || (I assume this is the hostname of the peer) | pattern | varchar(80) | | | || (Regex matching pattern?) | substitute | varchar(80) | | | || (Something else to do with regex substitution? Example?) | description | varchar(80) | YES | | NULL|| (Description of the entry) +-+--+--+-+-++ mysql describe rate; +--+--+--+-+-++ | Field| Type | Null | Key | Default | Extra | +--+--+--+-+-++ | rate_id | int(10) unsigned | | PRI | NULL| auto_increment | | route_id | int(10) unsigned | | MUL | 0 || | iso | char(2) | | | || (?? Description please? Example?) | type | char(3) | YES | | NULL|| (Type.. ? Example?) | country | varchar(40) | | | || (I guess this is freeform..) | extra| varchar(40) | YES | | NULL|| (Extra what?) | prefix | varchar(10) | | MUL | || (Ok, prefix of the number...) | active_date | date | YES | | NULL|| (Starting date of rate) | expires_date | date | YES | | NULL|| (Ending date of rate) | firstperiod | int(10) unsigned | | | 0 || (Explain?) | periods | int(10) unsigned | | | 0 || (Examples?) | startcost| int(10) unsigned | | | 0 || (Connection fee?) | periodcost | int(10) unsigned | | | 0 || (Cost per unit?) +--+--+--+-+-++ Once I get these question marks removed, I think it will be a very nice product. Thank you for your help! Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] default music source for SIP channel
The wiki says this about the MusicOnHold command: Plays hold music specified by class. If omitted, the default music source for the channel will be used. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold How do I set the default music on hold class for the SIP channel ? I tried adding musiconhold=test to my sip.conf. musiconhold.conf looks like this: [classes] default = quietmp3:/var/lib/asterisk/mohmp3 loud = mp3:/var/lib/asterisk/mohmp3 random = quietmp3:/var/lib/asterisk/mohmp3,-z test = quietmp3:/var/lib/asterisk/mohmp3,-z in extensions.conf I did: exten = 6000,1,Answer exten = 6000,2,MusicOnHold When I dial 6000 from a SIP phone ( xlite), musiconhold starts to play, but from the 'default' class. What am I screwing up ? -Lance -= Info about application 'SetMusicOnHold' =- [Synopsis]: Set default Music On Hold class [Description]: SetMusicOnHold(class): Sets the default class for music on hold for a given channel. When music on hold is activated, this class will be used to select which music is played. Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP and error talking to voicemail
Original Message Subject: RE: [Asterisk-Users] SIP and error talking to voicemail From: Dave Cotton [EMAIL PROTECTED] Date: Fri, January 09, 2004 1:03 am To: Asterisk List [EMAIL PROTECTED] On Fri, 2004-01-09 at 06:37, [EMAIL PROTECTED] wrote: How come every time I try connecting to their TFTP server I get permission denied? Something I'm doing wrong? tftp connect 130.94.123.253 tftp get bootload.bin Error code 2: Do not have permission to use this TFTP server I put the tftp address into my Grandstream and powered down/up et voila! Somewhere else to download 1.0.4.30 and 1.0.4.17 (just as a backup -- what I have now)? the http address has 1.0.4.18, 1.0.4.26 and 1.0.4.30 in zip form. -- Dave Cotton [EMAIL PROTECTED] Late night. I've been to http://www.grandstream.com/TEMP/FIRMWARE/ I just would like to find 1.0.4.17 so I know I'm not introducing any new bugs if I have to go back. I meant to say if you know somewhere else to get 1.0.4.38. I also tried just downloading it from my grandstream but it didn't seem to even want to try it -- probably the same problem. I still get permission denied when I try to TFTP manually also. hmm... If anyone has either of them, I'd appreciate a copy! Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] USA dial plan
Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with 1 in order To successfully make a call to other USA destinations? I have not been to USA (yet) :) Ta SJ In all cases of long distance, 1 plus the area code is used. In small areas where local only is involved you usually only dial 7 digits. In metro areas with multiple area codes, you use 10 digit dialing. Some places you use 10 digit dialing or 1 + area code, depends on the phone company.I've seen this happen on the east coast. Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP and error talking to voicemail
130.94.123.253 came from SIPphone not Grandstream, but even http://www.grandstream.com/TEMP/FIRMWARE/ only has 1.0.4.30 The only thing I can say is it's cleared my problems, making my GS usable again. -- Dave Cotton [EMAIL PROTECTED] How come every time I try connecting to their TFTP server I get permission denied? Something I'm doing wrong? tftp connect 130.94.123.253 tftp get bootload.bin Error code 2: Do not have permission to use this TFTP server Somewhere else to download 1.0.4.30 and 1.0.4.17 (just as a backup -- what I have now)? Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Grandstream Early Dial
On the config webpage, its on the bottom. Kevin Original Message Subject: Re: [Asterisk-Users] Re: Grandstream Early Dial From: Aaron Martin [EMAIL PROTECTED] Date: Sun, January 04, 2004 3:49 pm To: [EMAIL PROTECTED] Where / how do I set DTMF payload type to 101? - Original Message - From: Josh Roberson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 01, 2004 3:17 PM Subject: RE: [Asterisk-Users] Re: Grandstream Early Dial I've never had early dial working, however, I resolved my multiple digit issue by simply putting both the GS phones and asterisk in INFO mode. This worked on both 10.0.3.81 firmware on the budgetone and the ATA286, as well as 10.0.4.30 firmware. I'm not saying I don't believe you, but doubelcheck your lines in asterisk to be dtmfmode=info and the gs devices are on SIP INFO method, and your DTMF Payload type is 101. Just my $.02 -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] expression parsing
Hi. I've noticed a problem with the expression parsing in Asterisk. If the variable
is not defined, I will get a parse error. Yeah, there are ways around it, but I would
think that it should return false if 0, null, or undefined. I would change it, but I
have no idea about bison and I only have very basic C skills.
There was a bug opened on this, and there was a valid work-around posted, but I would
think that it would be 'nicer' if it would evaluate it this way. (Ref:
http://bugs.digium.com/bug_view_page.php?bug_id=401 ) If you put a 0 after the }
it does work as I would want it to without an error. The other suggestions did not
work. I propose for this bug to be re-opened.
extensions.conf:
exten = 1234,3,GotoIf($[${a}]?4:5)
If a is undefined:
WARNING[37910]: File ast_expr.y, Line 346 (ast_yyerror): ast_yyerror(): syntax error:
parse error
-- Executing GotoIf(SIP/1240-5eb6, 0?4:5) in new stack
-- Goto (default,1234,5)
If I change the extention to exten = 1234,3,GotoIf($[${a}0]?4:5) it works as expected.
Also, I'm not sure if this is my bad or what. If I use exten =
1234,3,GotoIf(${a}?4:5) and a is undefined:
-- Executing GotoIf(SIP/1240-2b8a, ?4:5) in new stack
-- Goto (default,1234,4)
it still returns true.
Behold the power of *,
Kevin
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RE: [Asterisk-Users] Problems with outgoing calls
I have a call generation script (very simple) to generate call load for testing, if that's what you're trying to accomplish. It's good for generating huge call volumes for IVR testing. Let me know if you need it! I would be interested in the script. If it is small, maybe you could post it to share on the list? Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Grandstream Quality Survey.... :P
Lubomir Christov [EMAIL PROTECTED] said: Yes, I know that the Grandstream firmware have problems (I have here 15 phones with some beta version already installed :( and waiting for bug fixing in the new beta) but the stable version 1.0.3.81 is working just perfect. Here too. Would be interested to learn what the problems are with 1.0.3.81. And if people complain about beta firmware, well... I guess that's why they call it beta, not? .. except that Grandstream are shipping new phones with the beta code ;-) Iain I just got 2 101s with 1.0.4.17 pre-installed which means I can't go back to 1.0.3.x. I really haven't had too many problems with it yet but I haven't used them much which I guess makes it bad. Shrug. Nothing major. Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
