similar to: Issue with using dialing PBX digits after call is connected

Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available Something like this would work: exten => 1234,1,Dial(SIP/providerA) exten => 1234,2,Dial(providerB) exten => 1234,3,Hangup But what if I want to put in a delay? If I put 30 seconds on each of them, I'll wait a

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

I'm running 1.2.9.1, and I can't get caller id dialplan matching to work. When calling from 9220370 to 1234, the following does not match. exten => 9220370/1234,1,NoOp(${CALLERIDNUM}) exten => 9220370/1234,2,Answer exten => 9220370/1234,3,Playback(tt-weasels) However, when calling from 9220370 to 1234, this DOES match. exten => 1234,1,NoOp(${CALLERIDNUM}) exten =>

hi folks. is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? suppose i want to do something like this: exten => 1234,1,dial(SIP/1234) exten => 1234,2,<do something> but when the dial command hangs up normally, line 2 won't get executed. -- Edwin Lam <edwin@officegeneral.com> Systems Engineer, Office

Hey all, I've setup * to serve the needs of our small helpdesk and I'm looking to expand. We're planning on doing support for different companies, each one identified by a different 1-800 number that terminates at our PBX. What I would like to know is: is there a variable I can read to determine what number any given caller dialed? I'd like to be able to separate calls based on

Hi, This is a bit off-topic, but still related to telephony. Is there a barebones TAPI driver that exists that would allow me to call up a command line with, as parameter, the number to dial. For exemple, Outlook integrates with TAPI, so that TAPI driver would allow me to call my own app with the phone number as argument. ex when clicking on 555-555-5555: the TAPI driver would call

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty

Hi All, I'm having trouble setting up a queue: I'm using AgentCallBackLogin to login in the queue, but: 1 - When an agent answer the call and another call arrive his phone rings again. 2 - When no there are no one answer the queue the system goes to voicemail of agent 1234 I'm using asterisk-1.2.0-beta1. My configuration is below, Any ideas? Many thanks, Joao Antunes

Thanks to this thread, we got it working too... but have a question... Once this is setup... does it stream forever, or does the stream only start when someone goes on hold/into a queue/etc? If it streams forever, at 24k... it looks like over 7GB/month in bandwidth... so we're not going to want to do that if a) it streams constantly and b) my math is correct. Thanks, Doug >

I can't seem to get the "r" modifier to work on inbound SIP calls. The way I understood this to work is that the channel would be answered, and a ring "tone" would be played to the channel. This is not very friendly in that it doesn't honor connection supervision rules, but... who cares? There are some instances where it may be in my interests to get a

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an

I'm stuck on queues! The way I read what documentation I have found, if I set up a queue like this: [general] persistentmembers = yes [testq] musiconhold=default strategy = ringall timeout = 10 retry = 5 context = testing member => SIP/100 and then add into extensions something like this: [incomingiax] exten => 1234,1,Dial(SIP/100,10) exten => 1234,2,Queue(testq|tTH|||300)

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hello everybody, I just buy a X101p clone and i'm new in asterisk. Here is my configuration : ISDN line ---- Alcatel ----PSTN ext 68-----Asterisk with X101p clone ------sip phone ext 200 - 203 ||| ISDN phones ext 60-67 >From sip phone to ext 60-67 it works. 9+extnumber >From sip phone to Land lines it works. 9+0+phone number >From ext

Here is what I had to change on the phone1.cfg file: I had this value in my 1.6.7 file, put in there following suggestions from the Wiki (http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501) : reg.1.server.1.expires="30" Now, this worked flawlessly with 1.6.7. But with 2.x, this seizes up the phone with a huge CPU load (approaching 100% at times) and makes it

Hi, I'm trying to call a SIP server while providing the SIP server username/password in dial string but it's not working ... Dial(SIP/gs102:test at 192.168.2.81); User on sip server (192.168.2.81): [gs102] disallow=all allow=ulaw allow=alaw type=friend username=gs102 secret=test host=dynamic dtmfmode=inband defaultip=192.168.2.1 qualify=1000 mailbox=102 context=context-gs102

Hello list, Can someone please point me out why would a stream like the following only write ONE line (the first) on the given file? Action: login Username: test Secret: 123456 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-000000: Append Cat-000000: default Var-000000: 127 Value-000000: >5555, Jason Bourne97, jason25 at noCia.gov.do ActionID:

There are often times that I want to read a DB value from the dialplan, and if this family/key pair does not exist, set it to some default value. for example: 1234,1 => Set(EMAILADDR=${DB(x/y)} 1234,2 => GotoIf($["${EMAILADDR}" = ""]?3:4) 1234,3 => Set(EMAILADDR=Someone@test.com) 1234,4 => NoOp(${EMAILADDR}) 1234,5 => Hangup() I have modified the db function

Hi, I`m getting this (from one of my registered phone that has been installed at some location I can`t access at the moment) in the Asterisk CLI. Notice the last 3 digits of the IP address in the error message: Apr 12 23:06:16 WARNING[25593]: chan_sip.c:7036 check_via: '67.39.117.611' is not a valid host Of course it's not a valid host! But, when using "sip show

Hi All I have an Asterisk box using a Sirrix card sitting between our PSTN and an ISDN pbx. Calls from the PSTN are forwarded to the PBX ok. Calls from the PBX are having problems - the digits being passed are being "garbled". The numbers from the PBX are totally incorrect and sometimes too long or too short. Anyone know what could be causing this? I would like to find some more