[Asterisk-Users] Santa Cruz, Bolivia?
I am in need of an Asterisk/Linux savvy person in Santa Cruz, Bolivia. Please contact me off-list. ___ 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] doublehash patch for 1.0.1
[EMAIL PROTECTED] wrote: is there a doublehash patch for 1.0.1? o old one to res_parking.c does not apply as there is no longer res_parking.c o wiki search is useless o google only finds the problems applying old patch to 0.7 I've attached an old-school, no frills, double-hash patch ported to the latest Stable with bug fixes CVS. Barton res_features.diff Description: Binary data ___ 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] doublehash patch for 1.0.1
[EMAIL PROTECTED] wrote: Just tried the patch you made with the latest CVS and it patches fine although it does not work. Now when I hit # it does not send the DTMF to the other side at all. Although hitting ## does get the transfer. Now # doesn't do ANYTHING :) I'm not sure why that is, it works with all our phones (Grandstream BT101s and analog phones on Grandstream ATA286s). I just tested by calling my bank's IVR. ___ 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] Disable flash hook hold?
[EMAIL PROTECTED] wrote: Currently, if I briefly press the flash hook on my phone, the caller is placed on hold. I would like for the channel to hangup if I do this instead, never placing a caller on hold (I'll be using call-parking instead). I disabled threewaycalling that is supposed to control this, but it doesn't make any difference: I'm assuming you mean zapata.conf not sip.conf. These settings do not affect sip clients. The sip client manages its own flash settings, not asterisk. Also, when you modify the zapata.conf file you must shutdown and restart asterisk for the changes to be recognized. I have the following settings in my zapata.conf and it works fine for me. [channels] callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=no echocancel=yes echocancelwhenbridged=yes echotraining=yes rxflash=50 The rxflash setting is to shorten the length of time the on-hook button needs to be depressed before a hangup is registered. Thanks. Yes, of course I meant zapata.conf, sorry. I've tried changing each of these settings: ; prewink: Pre-wink time (default 50ms) ; preflash:Pre-flash time (default 50ms) ; wink:Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ; rxflash: Receiver flashtime (default 1250ms) ; debounce:Debounce timing (default 600ms) Even when I set rxflash=1 and restart Asterisk, the flash-hold is still enabled... it might shorten the amount of time before hangup, but what I really need is to completely disable the flash-hold and hangup as soon as the flash is received. Perhaps this has something to do with another parameter such as debounce? Is there any documentation that describes in depth what those parameters actually do? ___ 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] Am I stupid or is my card DOA.?
[EMAIL PROTECTED] wrote: On 11-Oct-2004, Alex Barnes wrote: I had/have exactly the same problem with my X100P / TDM400P dev setup. I'm also having exactly the same problem with a TDM400P I received yesterday. I'm starting to suspect that seeing it work after swapping PCI slots is a placebo effect. Without moving the card it randomly seems to work about half the time. I have yet to successfuly get Asterisk to utilize the card, even on boots where the modules successfully load, but that might be misconfiguration on my part. I plan to call Digium for help either today or tomorrow. I, too, am having the same problem with a TDM400P that I received yesterday. It worked for a few hours, and then was no longer detected by the wcfxs.o module. No amount of rebooting solved the problem, and I replaced the card with an older TDM400P (with a defective port 1) and it was detected just fine. I have email Digium technical support, but have not heard anything back yet. ___ 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] TDM01B Goes missing after reboot
[EMAIL PROTECTED] wrote: On Oct 12, 2004, at 7:38 PM, Ian D. Wlloughby wrote: Hi All, I have just installed a TDM01B to fix my UK callerid and echo problems. In this respect everything is wonderful, however when I reboot wcfxs fails to load due to No Device found. If I power off and on everything is fine. I noticed that wctdm does not appear in /proc/interrupts after the reboot but does after power off/on. This seems similar to other peoples problems, do I have a duff card (Revision H) or is this a bug in wcfxs ? Regards Ian Ian, I responded to a similar posting today. With any luck, this workaround will also work for you. http://lists.digium.com/pipermail/asterisk-users/2004-October/ 067004.html Niles The exact same thing is happening to me. I received a response from Digium technical support this evening, and this is what they said: It is not a bad card, it is a new revision (Rev H), we are working on a fix. Sorry for your troubles. ___ 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] Disable flash hook hold?
Trying again with a different subject... Currently, if I briefly press the flash hook on my phone, the caller is placed on hold. I would like for the channel to hangup if I do this instead, never placing a caller on hold (I'll be using call-parking instead). I disabled threewaycalling that is supposed to control this, but it doesn't make any difference: threewaycalling: If enabled, you can place a call on hold by pressing a hook flash, whereupon you get a dialrecall tone and can make another call. Default: no. Here are the relevent sip.conf statements. What am I doing wrong? [channels] callwaiting = no cancallforward = no callreturn = yes immediate = no callwaitingcallerid = no threewaycalling = no transfer= no echocancel = yes echocancelwhenbridged = yes echotraining= 800 adsi= no busydetect = yes busycount = 8 callprogress= no musiconhold = random relaxdtmf = yes usedistinctiveringdetection=no useincomingcalleridonzaptransfer=yes ___ 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] Asterisk as Internet Talk Radio PBX system
James Sutton wrote: I see in the archives a brief thread between Barton and w last November 2003 about streaming to the Internet. I'd like to use an Asterisk to mediate multiple VOIP calls originated from the Internet to the studio to be mixed then passed out to an encoding PC thence back to Internet I am working on this very thing at the moment, although on a single box. Some specifics so far: Using Icecast/Ices to stream ogg. The stream is connected to a MeetMe on-air conference room. Music or whatever is connected from the line-in jack on the sound card using alsa and ices. Callers can come in VOIP or PSTN and are placed in hold conference room until they are bridged into the on-air room. Callers on hold listen to the live stream by using an application OggPlayer (a modified MP3Player application) that connects ogg123 into their room. I'm not currently doing any line-out (such as you want to send to your mixer), but plan to do so. I wonder if there is a group discussion of this type of functionality. Perhaps we've just started one :) Barton ___ 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] TDM400P and AGGRESSIVE_SUPPRESSOR dropping calls
Hi, I had been using 4 X100P cards in my Asterisk box, but 2 of them were sharing an interrupt. Therefore, periodically I would hear beeps and clicks that I had assumed were a result of this. So, I ordered a TDM400P with 4 FXO modules and installed it in the box last night. Today, we've had nothing but problems with it dropping calls. I installed the latest CVS of everything, and we've been getting random hangups. If I disable AGGRESSIVE_SUPPRESSOR, the random hangups seem to stop but we of course experience *really* bad echo. I have busydetect=yes and busycount=8, which has previously been working just fine with the X100Ps. Does anyone have an idea what's going on or how to fix it? ___ 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] TDM400P and AGGRESSIVE_SUPPRESSOR dropping calls
[EMAIL PROTECTED] wrote: I installed the latest CVS of everything, and we've been getting random hangups. Bruce Komito wrote: I, too, have a TDM400P with FXO cards and I am having the same problem. After further investigation, I thought that I had a bad module in the #1 position on my TDM400P. However, I just spoke with Digium technical support and they told me that there is a known problem with the #1 module position on the TDM400 card and that they are currently working on a software fix. They helped me configure an X100P to take the #1 module's place for now. ___ 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] asterisk-doc Conference Call - phase 2 :)
[EMAIL PROTECTED] wrote: Broadcast with app_ices to a shoutcast server For the world to listen too :P Has anyone gotten that app_ices to accually work? I sure as hell didn't. Yes, it works. Which part are you having problems with? Can you stream something with Icecast? Which config files do you want to see? ___ 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] Budgetone iLBC to IAX2 iLBC
[EMAIL PROTECTED] wrote: Where i can find this new firmware? Usualy i can download from http://www.grandstream.com/BETATEST/ but i only the stable version.. Thanks in advance Dimitri http://tinyurl.com/23s6m ___ 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] cron job to reboot GS101
[EMAIL PROTECTED] wrote: I know that you can reboot the GS phones by hitting the rs.htm URL on the phone. But, you have to log in to the web interface before doing this. I've attached a php script (quick and dirty hack) that resets the specified Grandstream devices. It requires the Snoopy class found here: http://snoopy.sourceforge.net/ resetgs.php Description: Binary data
RE: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
[EMAIL PROTECTED] wrote: exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew Yes the problem is that when making outgoing calls, there is enough of a delay in the call setup once the remote side picks up, that people that answer the phone hello will be heard saying o or if they talk fast enough not heard at all therefor leaving a very awkward silence at the start of a call. This is very annoying. A earlier person suggested answering the calls before dialing and playing a ringing sound till the start of the voice. That may be a work around of sorts for some, you will hear a ring then a congestion tone on call that can't connect, or a ring before a operator messages (say to dial one before the number) that most users may not be used to. I'll be playing with that ideal to see what odd effect a ring has before call setup causes. The work around may be less annoying then the problem. smile I'll see. I've seen the same thing, and it appears to be from attempting a native bridge. You can try the attached patch to disable native bridging. It cut out the annoying silence completely for me. This may be a bad thing (unnecessary CPU utilization due to same-codec translation), but I have not experienced any problems. Barton channel.c.diff Description: Binary data
RE: [Asterisk-Users] Cannot use # key to transfer calls
[EMAIL PROTECTED] wrote: I cannot use the # key to transfer a call. I have two kinds of SIP phones, Grandstream and IpDialog, and the # key cannot be used to transfer on either one. If I press the # key during a call, I hear the touchtone for it, but Asterisk does nothing. The documentation for parking a call says that I must first transfer the call using #, so that's why I need this feature to work. Thanks for any pointers. -Ron Dutt Make sure your Dial() line contains the 'T' and/or 't' options. Also make sure that your DTMF entries in sip.conf match the phones. I've found that with Grandstream HandyTones, the only reliable method of using '#' to transfer is by using inband DTMF, which means using ULAW/ALAW as well. ___ 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] IAXTel multiple registers?
With entries in sip.conf, I can route incoming SIP calls with an extension specified in the register command: register = user:[EMAIL PROTECTED]/123 The register command in iax.conf does not support specifying the extension. If I want to register multiple IAXTel accounts, how can I make them branch to different extensions or contexts when a calls arrives? ___ 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] IAXTel multiple registers?
Both register commands register with the iaxtel provider. No matter which number is dialed to reach Asterisk, it takes you to the same [provider] section, and thus the same context. I need for 2 register commands, registering to the same provider, to branch to different contexts or extensions. [EMAIL PROTECTED] wrote: You do this with contexts attached to the [provider] section in the iax.conf file and you provide coresponding contexts and extensions in your extensions.conf file. John Barton Hodges wrote: With entries in sip.conf, I can route incoming SIP calls with an extension specified in the register command: register = user:[EMAIL PROTECTED]/123 The register command in iax.conf does not support specifying the extension. If I want to register multiple IAXTel accounts, how can I make them branch to different extensions or contexts when a calls arrives? ___ 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 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 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: GS HandyTone-286 Transfer Problem, can anyone confirm?
[EMAIL PROTECTED] wrote: I have no problem transfer from one GS adaptor to another GS adaptor. /Hans-Henrik Andresen Can anyone confirm that this problem exists? The problem I'm experiencing with many GS adapters, regardless of firmware version is this. Call from one phone to another phone using both the 'T' and 't' flags in the Dial() command. After they are connected, you should be able to press '#' on either phone to hear transfer. What I am experiencing is the calling GS adapter will hear transfer when they press '#', but when the receiving GS adapter presses '#', nothing happens at all. Are you able to repeat this? If not, can you please tell me the firmware revisions and Asterisk version that you are using? Thank you very much. ___ 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] GS HandyTone-286 Transfer Problem, can anyone confirm?
There seems to be a problem related to the Grandstream HandyTone-286. When a call is placed through the adapter, the call can be transferred. However, when a call is received through the adapter, the call cannot be transferred. The problem does not exist with a BudgeTone-101 (1.0.4.23) using the same Asterisk configuration and Dial() settings (Ttm). I tried all of the firmware on their BETA site, from 1.0.4.35 through 1.0.4.50 and the problem was never solved. Can anyone confirm that this problem exists? Can anyone recommend an alternative analog telephone adapter that is in the price range of the HandyTone, but is actually reliable? Thank you. ___ 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] Zap to SIP transfer problem
I'm having a problem with transferring a call that comes in a Zap channel and is connected with a SIP channel (on a GS HT-286). The call is answered automatically, then the user enters an extension. Dial() is called with both T and t flags. When the bridge is made between the channels, the caller on the Zap channel can hit '#' to transfer, but the caller on the SIP channel cannot. No messages whatsoever are displayed on the console when the SIP user hits any keys. What am I missing? ___ 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] Zap to SIP transfer problem
I'm using SIP INFO and ulaw. It seems that the same thing happens from SIP to SIP as well, regardless of what the DTMF setting is. The actual problem is that the calling user can transfer, but the called user cannot. I just tried the latest CVS snapshot and the v1.0 stable branch and they both behave the same way. [EMAIL PROTECTED] wrote: Maybe you are using inband DTMF with a compressed codec. DTMF on a call with any codec other than ulaw or alaw MUST use OOB DTMF like RFC2833 or INFO. On Fri, 2004-03-05 at 20:39, Barton Hodges wrote: I'm having a problem with transferring a call that comes in a Zap channel and is connected with a SIP channel (on a GS HT-286). The call is answered automatically, then the user enters an extension. Dial() is called with both T and t flags. When the bridge is made between the channels, the caller on the Zap channel can hit '#' to transfer, but the caller on the SIP channel cannot. No messages whatsoever are displayed on the console when the SIP user hits any keys. What am I missing? ___ 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 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] Zap to SIP transfer problem
exten = s,10,Dial(${ARG1}/${DIALED},19,Ttm) which translates to
Dial(SIP/210-80f2, SIP/280|19|Ttm)
I believe the problem is related to the Grandstream HandyTone-286. A
caller can transfer, but a callee cannot. The problem does not exist
with a BT101 (1.0.4.23). I just tried all of the firmware on their
BETA site, from 1.0.4.35 through 1.0.4.50 and the problem was never
solved.
Can anyone confirm this for me?
I am SO SICK of dealing with HT-286 firmware bugs!
[EMAIL PROTECTED] wrote:
What is your ACTUAL Dial line?
On Fri, 2004-03-05 at 21:19, Barton Hodges wrote:
I'm using SIP INFO and ulaw. It seems that the same thing happens
from SIP to SIP as well, regardless of what the DTMF setting is.
The
actual problem is that the calling user can transfer, but the
called
user cannot. I just tried the latest CVS snapshot and the v1.0
stable branch and they both behave the same way.
[EMAIL PROTECTED] wrote:
Maybe you are using inband DTMF with a compressed codec. DTMF on a
call with any codec other than ulaw or alaw MUST use OOB DTMF like
RFC2833 or INFO.
On Fri, 2004-03-05 at 20:39, Barton Hodges wrote:
I'm having a problem with transferring a call that comes in a Zap
channel and is connected with a SIP channel (on a GS HT-286).
The call is answered automatically, then the user enters an
extension. Dial() is called with both T and t flags. When the
bridge is made between the channels, the caller on the Zap
channel
can hit '#' to transfer, but the caller on the SIP channel
cannot.
No messages whatsoever are displayed on the console when the SIP
user hits any keys. What am I missing?
___
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[Asterisk-Users] Stream both sides of conversation out sound card?
How feasable is it to get the Monitor app to combine the channels in pseudo-real-time and have the resulting audio stream out a soundcard? ___ 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] Integrating with an existing PBX
I have successfully setup several standalone Asterisk systems and they work great. I have the opportunity to integrate Asterisk with an existing Toshiba CHSUB672A PBX. I believe that the way I should connect the systems is through a T1 interface on the Toshiba, and a T100P on the Asterisk box like such: |-| |--| | Toshiba PBX |(T1)---(T100P)| asterisk |(eth0)---(Wireless)---Phone |-| |--| Does this seem correct? I am not familiar with proprietary PBX systems, and I need to determine if the Toshiba has a T1 interface. Their website describes the CTX670 as having one but does not have any information about the CHSUB672A. What is the best avenue for obtaining additional information about the PBX (such as it's features and configuration methods) since I don't have any documentation on it? Thanks a bunch, Barton ___ 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] Cause of transfer problem (GRANDSTREAM!)
It turns out that the cause of the transfer problem is the Grandstream 1.0.4.39 firmware. I was shipped a bunch of HandyTone-286 devices that contained the 1.0.4.30 firmware. This version had a bug where the phone would sometimes not ring at all. I was told by Grandstream to upgrade to the 1.0.4.39 version. This broke the Use # as Dial Key option, and evidently transfer as well. I still do not have any problems with my 1.0.3.81 phones, but I've read that I cannot downgrade from a 1.0.4x version to a 1.0.3x version. I'm pretty pissed that they shipped me what I consider to be defective devices, do not give me a way to back down to a usable version, and do not have a fix for this problem that makes all of the devices completely unusable to me. ___ 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] Grandstream transfer solution + DTMF translation possible?
The solution to the problems with the Grandstream 1.0.4.39 firmware is to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to work. However, this presents another problem. When I'm using g729 to place a call, I get the warning Unable to process inband DTMF because inband is not supposed to work with g729 (although it does seem to work when I've tried it so far). Can Asterisk convert between different modes of DTMF? For instance, my phone would use inband, and Asterisk would convert this to rfc2833 before reaching the channel I am connected to? I've tried using dtmfmode=rfc2833 in the service definition in sip.conf. Thanks, Barton ___ 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] Transfer problem
Is anyone else experiencing problems with Transfer via # and the 'T' or 't' flags passed to Dial()? I've tried both the latest CVS and 0.7.1 tarball. If I dial in from a pstn line and then choose an extension that dials a SIP phone with Ttm flags, when I press # on the SIP phone, the pstn caller hears the Transfer and the SIP phone gets the music on hold. I can't make the SIP phone initiate a transfer by pressing #, no matter what I try. Barton ___ 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] Packet8 DTA310 Advanced Configuration
Hi, I just received a DTA310 Terminal Adapter from Packet8. The Advanced Configuration is password protected. Does anyone know the default password or algorithm necessary to get into it? Thank you, Barton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A solution to free line notification
Barton Hodges wrote:
I've been messing around with a free line notification
where an extension is dialed for a second when a line becomes
available. I can't seem to get the h extension to continue
when the local party hangs up. I've seen references to other
people having the same problem in the list archives, and the
solution presented was to use AGI.
I finally figured out how to get this to work. Thanks to one of Steven
Critchfield's emails today, I found out about sample.call and
/var/spool/asterisk/outgoing which is what I needed to control the
dialing. It seems that if you call a macro from within an h
extension, only one, or a few select lines get called before the macro
returns. I messed around with different alternatives until I found one
that worked. I would give anything for control structures and
user-defined functions within the dialplan. A nice little for() loop
would tidy things up nicely. Is AGI what I need to be using? I wasn't
sure how to do things such as Dbget(), except through the Exec() call.
Here are snippits to show how it was done:
/var/lib/asterisk/agi-bin/fln.agi:
#!/bin/sh
[ $# -gt 0 ] || exit 0;
echo -e Channel: ${1}
WaitTime: 1
Callerid: Free Line Notification (000) 000-
Context: default
Extension: s
Priority: 1 /var/spool/asterisk/outgoing/fln.$$
/etc/asterisk/extensions.conf:
[from-inside]
include = to-internal
include = app-freeline
exten = h,1,Macro(hangup)
[check-fln]
exten = s,1,DBget(TECH=FLN/${EXT})
exten = s,2,ChanIsAvail(Zap/1Zap/2Zap/3Zap/4)
exten = s,3,DBdel(FLN/${EXT})
exten = s,4,AGI(fln.agi,${TECH}/${EXT})
exten = s,5,Goto(macro-hangup,s,${PRI})
exten = s,102,Goto(macro-hangup,s,${PRI})
exten = s,103,Goto(macro-hangup,s,${PRI})
exten = s,104,Goto(macro-hangup,s,${PRI})
[macro-hangup]
exten = s,1,SetVar(PRI=4)
exten = s,2,SetVar(EXT=111)
exten = s,3,Goto(check-fln,s,1)
exten = s,4,SetVar(PRI=7)
exten = s,5,SetVar(EXT=112)
exten = s,6,Goto(check-fln,s,1)
exten = s,7,SetVar(PRI=10)
exten = s,8,SetVar(EXT=113)
exten = s,9,Goto(check-fln,s,1)
exten = s,10,Wait(1)
exten = s,11,Hangup
[macro-goodbye-hangup]
exten = s,1,Playback(vm-goodbye)
exten = s,2,Macro(hangup)
[app-freeline]
exten = _*98,1,Cut(CHAN=CHANNEL,-,1)
exten = _*98,2,Cut(TECH=CHAN,/,1)
exten = _*98,3,Cut(EXT=CHAN,/,2)
exten = _*98,4,DBput(FLN/${EXT}=${TECH})
exten = _*98,5,Answer
exten = _*98,6,Playback(contrib/activated)
exten = _*98,7,Playback(vm-for)
exten = _*98,8,Playback(vm-extension)
exten = _*98,9,SayDigits,${CALLERIDNUM}
exten = _*98,10,Macro(goodbye-hangup)
exten = _*99,1,Cut(CHAN=CHANNEL,-,1)
exten = _*99,2,Cut(EXT=CHAN,/,2)
exten = _*99,3,DBdel(FLN/${EXT})
exten = _*99,4,Answer
exten = _*99,5,Playback(contrib/de-activated)
exten = _*99,6,Playback(vm-for)
exten = _*99,7,Playback(vm-extension)
exten = _*99,8,SayDigits,${CALLERIDNUM}
exten = _*99,9,Macro(goodbye-hangup)
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RE: [Asterisk-Users] Prefix the * character
[EMAIL PROTECTED] wrote:
I have a smilar problem : I have a default context for an interface,
where I'd like to prefix all incoming calls DID numbers (basically,
the telco sends the last 4 digits dialed, I want to fully qualify my
E164 number before doing extensions processing).
I don't know much (yet!) about Asterisk, so I thought something like
exten = s,1,Prefix(3312345)
include = my_local_e164_extensions
would do the trick. Unfortunatly, if the ${EXTEN} was 6060 at that
time, I get a new extension as s6060 (instead of 33123456060). Is it
supposed to be this way ?
So instead I had to do something like
exten = _,1,Prefix(3312345)
include = my_local_e164_extension
which works fine, except that now I'm at the 2 level in the
context,
and I had to modify my_local_e164 extension context accordingly.
Does somebody know of a better way to do it ?
Thanks.
The lines in a context get reordered. If you want to force the order
of those lines, put the exten lines in separate contexts and include
them... something like this:
[some-context]
include = prefix
include = my_local_e164_extension
[prefix]
exten = _,1,Prefix(3312345)
I don't know if that will solve your problem, but it is something to
consider.
Barton
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RE: [Asterisk-Users] Prefix the * character
[EMAIL PROTECTED] wrote:
On Mon, Dec 08, 2003 at 08:58:07AM -0600, Barton Hodges wrote:
The lines in a context get reordered. If you want to force the
order
of those lines, put the exten lines in separate contexts and
include them... something like this:
[some-context]
include = prefix
include = my_local_e164_extension
[prefix]
exten = _,1,Prefix(3312345)
I don't know if that will solve your problem, but it is something
to
consider.
My problem is that the exten lines in my_local_e164_extension
still
have to start at 2, since prefix used the 1 position, and that's
what I'd like to avoid by using s.
To do that, I put immediate=yes on my PRI in zapata.conf, but
unfortunatly the Prefix command will use s as the extension, and
generate a new extension like 3316918s, which is not really nice.
Is there any way ta manipulate ${EXTEN} as a variable, rather that
wich the Prefix function ? If so, I haven't found out.
You could always do something like this:
exten = 2,1,Dial(WHEREVER,3312345${EXTEN})
or assign it to a new variable:
exten = 2,1,SetVar(NEWEXTEN=3312345${EXTEN})
exten = 2,2,Dial(WHEREVER,${NEWEXTEN})
If I still don't have a grasp on what you're trying to accomplish,
could you post your extensions.conf?
Barton
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[Asterisk-Users] Dial T option not obeyed with Grandstream BT101
In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register = number:[EMAIL PROTECTED] extensions.conf: [from-sip] exten = s,1,Dial(SIP/111SIP/117) exten = 111,1,Dial(SIP/111,20) exten = 117,1,Dial(SIP/117,20) 1. The calling user dials number, which drops them into [from-sip] 2. Extensions 111 and 117 are Dialed. 3. The called user picks up extension 111. 4. The calling user presses Transfer on the Grandstream phone, then dials 117 and presses Send. 5. The called user on extension 111 is then transferred to extension 117. I don't believe this is supposed to happen because I have not specified the T option to the Dial command. Even without any options specified at all, both the calling and called users are able to transfer the call. I'm using a CVS snapshot from Sun, Nov 30th 04:04:45, 2003. What am I missing here? Barton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to find path from G729A to ULAW on Sipphone.com
[EMAIL PROTECTED] wrote: I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A Before somebody tells me UTFG, I ALREADY HAVE. Somebody else had a similar issue last week and there was no real resolution posted. So here it is again. I have all of the codecs that I support enabled in my sip.conf. Here is the relevant section: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls srvlookup = yes ; Enable SRV lookups on outbound calls pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw ; Allow codecs in order of preference allow=gsm allow=ilbc register = 17476692375:[EMAIL PROTECTED]/1101 [sipphone] type=peer username=17476692375 secret=[MYSECRET] host=proxy01.sipphone.com fromuser=SteveSokol fromdomain=sipphone.com canreinvite=no ; ==END OF SIP.CONF FILE=== The issue occurs whenever any calls that route over the sipphone peer are made to a toll-free number. The calling phone (either my GS100 or my X-LITE softphone) rings two or three times then gives me busy. Here is the entire debug output: -- Executing Dial(SIP/1101-1f83, SIP/[EMAIL PROTECTED]|20|tr) in new stack -- Called [EMAIL PROTECTED] NOTICE[1234379840]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1234379840]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A -- SIP/sipphone.com-e7b3 is making progress passing it to SIP/1101-1f83 -- SIP/sipphone.com-e7b3 answered SIP/1101-1f83 -- Attempting native bridge of SIP/1101-1f83 and SIP/sipphone.com-e7b3 NOTICE[1242768320]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1242768320]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A WARNING[1234379840]: File chan_sip.c, Line 1159 (sip_write): Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) == Spawn extension (default, 918884510851, 1) exited non-zero on 'SIP/1101-1f83' The problem does NOT occur when I call another sipphone.com user (i.e. GS100 - Asterisk - Sipphone - GS100). Those calls go through just fine. The toll free calls were working last week. Is it me, or is it Sipphone.com? Any suggestions would be greatly appreciated. Steve I've been having the same types of problems (I'm probably the guy you're referring to who had the same problems last week). This is the solution I have found to work reliably so far. Configure the Grandstream BT101 with the following codecs, in the following order: choice 1: G.729A/B (g729) choice 2: PCMU (ulaw) choice 3: PCMA (alaw) choice 4: G.729A/B (g729) choice 5: PCMU (ulaw) choice 6: PCMA (alaw) Configure the codecs in sip.conf like this: disallow=all allow=all allow=ulaw allow=alaw allow=g729 Configure the entry in extensions.conf to use a certain codec when necessary (I've found it necessary only when calling through the 800 gateway provided to both FWD and SIPphone): ; FWD exten = _1800NXX,1,Macro(callerid-pstn) exten = _1800NXX,2,SetVar(SIP_CODEC=g729) exten = _1800NXX,3,Dial(SIP/[EMAIL PROTECTED]) ; SIPphone ;exten = _1800NXX,1,Macro(callerid-pstn) ;exten = _1800NXX,2,SetVar(SIP_CODEC=g729) ;exten = _1800NXX,3,Dial(SIP/[EMAIL PROTECTED]) I hope this helps, Barton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to find path from G729A to ULAW onSipphone.com
[EMAIL PROTECTED] wrote: i followed what you said didint work heres what console says i cant do the 1800 call anyway -- Executing Macro(SIP/101-8376, callerid-pstn) in new stack -- Executing SetVar(SIP/101-8376, SIP_CODEC=g729) in new stack -- Executing Dial(SIP/101-8376, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/fwd-2e46 is making progress passing it to SIP/101-8376 -- SIP/fwd-2e46 answered SIP/101-8376 == Spawn extension (asterisk, 18006927753, 3) exited non-zero on 'SIP/101-8376' -- Executing Macro(SIP/101-c43c, callerid-pstn) in new stack -- Executing SetVar(SIP/101-c43c, SIP_CODEC=g729) in new stack -- Executing Dial(SIP/101-c43c, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/fwd-bc38 is making progress passing it to SIP/101-c43c -- SIP/fwd-bc38 answered SIP/101-c43c == Spawn extension (asterisk, 18006927753, 3) exited non-zero on 'SIP/101-c43c' You need to modify the lines in extensions.conf to match your configuration: Try this: exten = _1800NXX,1,SetVar(SIP_CODEC=g729) exten = _1800NXX,2,Dial(SIP/[EMAIL PROTECTED]) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Streaming channels from Asterisk to the Internet
Hi folks, I'm wondering if it is currently possible to configure Asterisk to forward the conversations from 2 channels into a streaming daemon, such as Icecast, so that other people connected to the Internet can listen. The concept is similar to a radio talk-show. The show host would connect to Asterisk via an X100P or through VOIP. His or her voice would then be sent to the streaming daemon for those on the Internet to hear. The show host would also have control of the other incoming channels (via a custom web-interface), which would come in via an X100P or VOIP as well. The show host and the chosen channel(s) could have a conversation streamed out to the Internet until the channel is disconnected by the host. Any input regarding the feasability of this, and the available software (such as asterisk-perl) that can be used to accomplish this would be greatly appreciated. Barton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Solution to dialing 800 numbers through FWD or SIPphone
Thanks to John Lodden's help, he was able to determine that the cause of my inability to dial 800 numbers through FWD or SIPphone was due to the Grandstream phone and the order of codecs in sip.conf This order breaks the 800 dialing: disallow=all allow=ulaw allow=alaw allow=g729 However, this order allows it: disallow=all allow=g729 allow=ulaw allow=alaw Barton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialing 800 numbers through FWD or SIPphone?
Barton Hodges wrote:
Hi,
Does anyone know how to dial toll-free (800) numbers through FWD or
Siphone?
Using the configuration below, I can dial out to SIPphone.com users
by simply dialing their number (1747XXX) and can dial out to FWD
users by dialing 1383FWD#
However, when I dial 18005551212 through SIPphone, or through FWD
(depending upon which line is selected in ; 800 Toll Free Numbers
below, I receive a 403 Forbidden response. From what I've read,
this might be due to outbound proxy authentication with FWD, but I
don't believe that SIPphone.com is using proxy authentication. I can
configure the phones to connect directly to both FWD and SIPphone.com
and they work when dialing the 800 numbers.
Any suggestions would be greatly appreciated.
Barton
---
sip.conf:
[fwd]
type=friend
username=FWD#
secret=secret
host=fwd.pulver.com
[sipphone]
type=friend
username=SIPPHONE#
secret=secret
host=proxy01.sipphone.com
fromuser=SIPPHONE#
fromdomain=proxy01.sipphone.com
extensions.conf:
; 800 Toll Free Numbers
exten = _1800XXX,1,SetCallerID(${CALLERIDNUM})
exten = _1800XXX,2,SetCIDName(${CALLERIDNUM})
exten = _1800XXX,3,Dial(SIP/[EMAIL PROTECTED])
;exten = _1800XXX,3,Dial(SIP/[EMAIL PROTECTED])
exten = _1800XXX,4,Hangup
; SIPphone.com
exten = _1747XXX,1,SetCallerID(${CALLERIDNUM})
exten = _1747XXX,2,SetCIDName(${CALLERIDNUM})
exten = _1747XXX,3,Dial(SIP/[EMAIL PROTECTED])
exten = _1747XXX,4,Hangup
; Free world dialup
exten = _1393.,1,SetCallerID(${CALLERIDNUM})
exten = _1393.,2,SetCIDName(${CALLERIDNUM})
exten = _1393.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _1393.,4,Hangup
[EMAIL PROTECTED] wrote:
You need to add # before dialing the rest of numbers.
Ta
SJ
Thanks for the response. If you mean changing the Dial line in
extensions.conf to something like this, then I really don't see how that
is correct, and it does not seem to work:
; 800 Toll Free Numbers
exten = _1800XXX,1,SetCallerID(${CALLERIDNUM})
exten = _1800XXX,2,SetCIDName(${CALLERIDNUM})
exten = _1800XXX,3,Dial(SIP/[EMAIL PROTECTED])
;exten = _1800XXX,3,Dial(SIP/[EMAIL PROTECTED])
exten = _1800XXX,4,Hangup
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RE: [Asterisk-Users] Dialing 800 numbers through FWD or SIPphone?
[EMAIL PROTECTED] wrote:
exten = _1800XXX,1,SetCallerID(${CALLERIDNUM})
That should be =
Ah yes, search and replace without forethought or inspection
to include my previous email indented with for informational
purposes. Alas, it does not work even with =.
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[Asterisk-Users] Dialing 800 numbers through FWD or SIPphone?
Hi,
Does anyone know how to dial toll-free (800) numbers through FWD or Siphone?
Using the configuration below, I can dial out to SIPphone.com users by
simply
dialing their number (1747XXX) and can dial out to FWD users by dialing
1383FWD#
However, when I dial 18005551212 through SIPphone, or through FWD (depending
upon which line is selected in ; 800 Toll Free Numbers below, I receive
a 403 Forbidden response. From what I've read, this might be due to
outbound proxy authentication with FWD, but I don't believe that
SIPphone.com
is using proxy authentication. I can configure the phones to connect
directly
to both FWD and SIPphone.com and they work when dialing the 800 numbers.
Any suggestions would be greatly appreciated.
Barton
---
sip.conf:
[fwd]
type=friend
username=FWD#
secret=secret
host=fwd.pulver.com
[sipphone]
type=friend
username=SIPPHONE#
secret=secret
host=proxy01.sipphone.com
fromuser=SIPPHONE#
fromdomain=proxy01.sipphone.com
extensions.conf:
; 800 Toll Free Numbers
exten = _1800XXX,1,SetCallerID(${CALLERIDNUM})
exten = _1800XXX,2,SetCIDName(${CALLERIDNUM})
exten = _1800XXX,3,Dial(SIP/[EMAIL PROTECTED])
;exten = _1800XXX,3,Dial(SIP/[EMAIL PROTECTED])
exten = _1800XXX,4,Hangup
; SIPphone.com
exten = _1747XXX,1,SetCallerID(${CALLERIDNUM})
exten = _1747XXX,2,SetCIDName(${CALLERIDNUM})
exten = _1747XXX,3,Dial(SIP/[EMAIL PROTECTED])
exten = _1747XXX,4,Hangup
; Free world dialup
exten = _1393.,1,SetCallerID(${CALLERIDNUM})
exten = _1393.,2,SetCIDName(${CALLERIDNUM})
exten = _1393.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _1393.,4,Hangup
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