similar to: Problem with outbound dialing from remote phone

Hi Guys, I have a need to alter the general timeout in Asterisk. I am wondering if this is something that is hard coded into Asterisk code or if there is a parameter that can be set somewhere. For outbound, I am using x. and hence unless I append a # sign, I would have to wait maybe 5 seconds or so for the call to go through. Is there anywhere in Asterisk that I can change this 5 seconds to

Dear list, on Asterisk 1.4.21 which is being used in a callthrough scenario - callers call via PSTN to a DID coming in via SIP and then dialing outbound via DTMF and the outbound calls get routed via some SIP termination provider - lately I see that every now and then MusicOnHold gets triggered like this on outbound calls: Started music on hold, class 'default', on

That was the issue, thanks. I now am able to get the caller ringing on an outbound call, but an external phone number (E164) I am dialing does not ring. On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <george.joseph at fairview5.com > wrote: > > > On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I have setup my

List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer

Hello, I am using the voicemail module of asterisk. When I did some test calls from my mobile phone, sometimes the beginning of the prompt was missing, e.g. instead of something like "number 12345 not available" I was only hearing "345 not available". Verbose level 5 on the asterisk console didn't give me any hint on this, it only shows that playback of the prompt started

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before

We have 2 PRI from AT&T And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised [Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request, cause 18 [Jun 7 17:57:10] DEBUG[24856] sig_pri.c: Not yet hungup... Calling hangup once with icause,

Yes, I think the dial does get executed (sonny calling outbound 202-555-1212): core set verbose 3 Console verbose was OFF and is now 3. -- Executing [912025551212 at from-internal:1] Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from "" <sonny> to 12025551212 through fromgw") in new stack [Mar 15 19:27:06] NOTICE[16648][C-00000022]: Ext. 912025551212:1 @

Does anybody know of a way to test whether a mySQL connection invoked from the dialplan is current or not? For example: extensions.conf =============== [context] exten => _X.,1,MYSQL(Connect connid localhost user pass db) exten => _X.,n,MYSQL(Query resultid ${connid} SELECT `something` FROM `table` WHERE `number` = ${EXTEN}) exten => _X.,n,MYSQL(Fetch foundRow ${resultid} something)

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic configuration works, and I am connected to a SIP trunk using SIP.US, and have set up my inbound calling which works correctly (when I call my PBX DID, the call does come into my PBX network). The issue is that I am not able to make outbound calls, because the call fails with the error:

I have a puzzling situation, and would be grateful for any insight. I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g. [from-siptrunk] exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) Now, if I use a different SIP trunk for the outbound call, than the inbound call came on, the call is set up

Hai, Does anybody have problems with a wrong Connected Line ID with asterisk version 1.6 The following bug was for version 1.4, but I cannot make up if this bug is still in version 1.6 http://forums.digium.com/viewtopic.php?t=7780 In version 1.8 it is possible to change the Connected Line ID, but this isn't the case in version 1.6 Regards, Arjan Kroon Mobillion BV

Hi Asterisk guys, Flite is not working with asterisk 1.6.2.17. Flite is working with asterisk 1.4. Please help me how to use it with asterisk 1.6 ....... Thanks in advance. ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

Can anyone recommend a good WebRTC phone to use with Asterisk? I do not mind if it is commercial or open source. Customers are starting to ask for web solutions and we need to start testing. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside

Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890