search for: opschil

Displaying 20 results from an estimated 22 matches for "opschil".

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2006 Apr 04
2
Can't get Pickup app working
I'm trying to set the Pickup feature. I'm setting my extensions.conf as: exten => _*.,1,Pickup(SIP/${EXTEN:1}) but if, for example, extension 03 is ringing by a call made from extension 01, and I try to pick it up from extension 02 (by dialing *03 from extension 02), I can see in the Asterisk console (Verbosity set to 10): -- Executing Dial("SIP/01-512c",
2006 May 23
3
Transfer extensions processing control to Manager
I'm developing an application that monitors the state of the incoming calls using Manager events. So, as a part of it, I need to "override" the control of the extensions by the dialplan itself. The problem is that, if I don't declare the incoming extension, Asterisk hangs up the call by default. So I want to know if there's some kind of "ManagerControl() application
2006 Mar 15
2
Speeding up the dial of DTMF's in SIP channel
I'm dialing DTMF's in a SIP channel using the options: [sip.conf] dmtfmode=info [extensions.conf] exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN})) (this is a custom SIP gateway, which receives the DTMF's sent from softphones through Asterisk, and based on them, build the destination PSTN number). My problem is that Dial send the DTMF's to the SIP/gateway user at a rate
2006 May 11
2
Problem setting locale for voicemail
I've set voicemail almost successfully, only a minor detail remains :-) I can't get the dates in my local language (spanish). In sip.conf, zapata.conf and voicemail.conf, I've set: language=es and my locale is "es" also. However, the days and months names still appear in english in the emails!!! Thursday 11 de May de 2006, 18:49:34. instead of Martes 11 de mayo de
2006 Mar 09
2
Extracting info from the $EXTEN variable
Is there a way to access only certain positions in the $EXTEN variable? I'd like to filter my international calls based on the destination country: My dialplan looks like this (1XX0. is the international calling convention for Chile) exten => _1XX0.,1,Dial(SIP/${EXTEN:4}@external_provider) But, I'd like to, depending on the destination country (digits 5 and eventually 6 of EXTEN),
2006 Feb 23
3
Codec order sent wrong from Asterisk
I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my
2006 May 10
2
Is there a way to not propagate a context included inside other context?
I've defined my dialplan as showed below. My internal lines are numbered as 12345XX, and internal users can call another by the entire 7-digits extension, or by just last 2 digits. [invalid] exten => _X.,1,Playback(pbx-invalid) exten => _X.,2,Hangup() [internal] include => invalid exten => _XX,1,Dial(SIP/12345${EXTEN}) ; Short alias for internal lines exten =>
2006 Feb 22
0
Is SIP "canreinvite" working ok?
I've the following situation: Phone A: Codec GSM supported Phone B: Codec iLBC supported in sip.conf: [general] ... disallow=all allow=gsm allow=ilbc allow=alaw allow=ulaw canreinvite=yes ... (There's a lot of other SIP users, that's why I made the default codec list bigger than just GSM and/or ALAW) If phone A calls to phone B the conversation is established at SIP level, but
2006 Feb 23
1
How can I force Asterisk t not override my codec order?
I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to softphone 2, I presume that Asterisk should do transcoding (canreinvite is set to no):
2006 Mar 06
0
Information to program a new driver for Asterisk
I'm interested in developing a new channel driver for a thrid party telephony card for Asterisk. Is there any "official" document that explains how to do this? We've been looking the doc/channel.txt and doc/modules.txt in the source, but that's not a very complete source of info :) Thanks a lot for your attention. -- Atly. Alvaro Palma
2006 Mar 07
1
Changing REINVITE status of the channel dynamically
I've an Asterisk server running in my office, which forwards all long distance calls to a third party SIP service using an extension rule: exten => _1XX0.,1,Dial(SIP/{EXTEN:4}@external_sip_server.com) (1XX0 is the international calls rule for Chile) Also, in my sip.conf, I've defined canreinvite=yes to decrease the network load to the server caused by the RTP. However, the external
2006 Mar 08
1
Upgrading Asterisk witk G729 license installed
I've an Asterisk 1.2.4 installation, where I've also installed the G729 codec license. I'd like to upgrade that installation to 1.2.5, but I'm not sure if I'll lost the license in the process (and if I'll be able to recover it later!!!). Is there any special consideration I've to keep in mind in this case, or should I just run the typical "make + make
2006 Mar 23
0
Problem with INVITE's being sent
I've being testing a couple of GrandStream ATA 286 which with no reason start responding 486 Busy to all new incoming INVITES. They are connected to an Asterisk installation as SIP client. Running ethereal between them, I could notice that, for some reason unknown for me at this time, Asterisk sends some stranges INVITE's AFTER the communication has been established and acknowledged
2006 Mar 30
0
Strange second REINVITE being sent
I'm using Asterisk a SIP Server for a lot of GrandStream HandyTone ATA's. Each one of them is configured in sip.conf as: [1234567] type=friend username=1234567 secret=1234567 callerid="ATA 1234567" host=dynamic nat=yes qualify=yes disallow=all allow=g729 canreinvite is set globally to YES. When one ATA calls another, I see the next traffic on Ethereal (just shown the sequence
2006 Apr 10
0
Problem with Asterisk and Grandstream HT286
I've dealing with this issue for a while, and I'd really like to know if anybody has experienced the same pain before :-) I've a lot of Grandstream HandyTone 286, loaded with the latest firmware (1.0.8.16) from the GS website. In my sip.conf, this ATA's are configured as: [05] type=friend username=05 secret=XXXX callerid="User 05" host=dynamic nat=yes qualify=yes
2006 May 09
0
How to distinguish between UNEXISTENT channels v/s UNAVAILABLE channels
Is there a way to distinguish, in the answer of the Dial command, when a channel is not available (for example, an unregistered but valid SIP user) v/s when the dialed channel is inexistent, even when it matches an extension? For example, I've the following simple dial plan: exten => _XX,1,Dial(SIP/${EXTEN},10,) exten => _XX,2,GotoIf(DIALSTATUS = CHANUNAVAIL?4:3) exten =>
2006 May 23
1
Problem with options to "Dial" application
I'm trying to set a dialing rule in my dialplan. As a part of it, from my point of view, this works wrong priorityjumping=no [test_context] exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With "j" flag exten => 1234,2,Playback(digits/2) exten => 1234,3,Playback(digits/3) exten => 1234,102,Playback(digits/4) In this case, if I dial the extension, and
2006 May 24
1
Generate two calls from Asterisk and bridge them
Is there a way in Asterisk (I guess there's, it's only I can't figure out how :-)) to: 1.- Generate a call to channel 1 (example, to PSTN v?a an E1 card, using Zap/g1) 2.- Generate a call to channel 2 (example, an internal SIP extension). 3.- Once both channel have answered, connect the call between them. This way, I can, for example, play audios in both channels before they are
2006 May 30
0
IAX softphone with RSA support?
Which (preferible free :-) softphone that supports IAX and RSA encryption do you recommend? It seems that IDEFisk doesn't yet. Thanks a lot for your help. -- Atly. Alvaro Palma
2006 Jun 02
0
Limiting the size of a Queue
Is there a way to limit the size of a Queue? I want to create a queue with for example, 5 agents, and only allow at most 10 persons waiting so this way, they don't saturate my entire PSTN span, which can be also simultaneously used for another Queues or for my outgoing calls. Thanks a lot for your attention. -- Atly. Alvaro Palma