similar to: budgetone problem on hangup

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

Hello list, my case is as follows: SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729. When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the keypad on the phone. As suggested by you, I need to configure the SIP1 with out band dtmf mode , what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238 ? do I also need to make same kind

Hi, I've seen some answers to this on the mailing list archives but nothing that seems like the right answer. What I want is for 2 SIP phones to use speex to talk to each other through 2 asterisk boxes (linked over IAX2) while only supporting ulaw on the asterisk boxes themselves. I think a diagram will help ;) SIP1 <--> *1 <--> IAX2 link <--> *2 <--> SIP2 I want

Grrr. I'm using outlook web access and there's no way to do inline replies. Anyway... Gabriel. Using SER does not create a single point of failure. You install three SER boxes. Single point of failure gone. It does not take several seconds. If your phones are configured for SRV, and 2/3 of your SER boxes down, it takes about 2s. That's not bad for a 2/3 system failure. You can

Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

Hey, You know, the Digium guys said both are good. They said the the DNS method is better because you dont have the extra point of failure (SER) but said the SER method is better in that it gives you more exact control in the handling of the calls and registration. They did acknowledge there would be a possible downtime only for incoming calls to users with dynamic IPs if the

Hi All I want to do features as belows. user ---> call ( from telco) --> asterisk ---> IVR -- SIP 1. after that, SIP1 transfer to SIP2 (unattendant or attendant transfer). i want to SIP1 hear stream sound data of call conversation between user and SIP 2 (don't used call conference) SIP3 want to hear stream sound data of user and SIP2 conversation, can be press DTMF

Hi All I want to setting as belows. caller ---> call ( from telco) --> asterisk ---> IVR -- SIP 1. after that, SIP1 transfer to SIP2 (unattendant or attendant transfer). i want to SIP1 hear stream sound data of call conversation between caller and SIP 2 (don't used call conference) SIP3 want to hear stream sound data of caller and SIP2 conversation, can be press DTMF

Hello, I am experiencing weird problem in Asterisk 16.2, possibly a bug. Same thing works fine in Asterisk 11. Here is the situation: I have 2 extensions on 2 phones. 4 extensions in total. phone 1: 8882 8382 phone 2: 8884 8384 And I have 2 SIP trunks for outgoing calls. I want to call via SIP1 when called via 8882 or 8884, and SIP2 when called via 8382 or 8384. And one last detail. SIP1

Dear all, I'd like to know if there is a way for multiple asterisk servers to share a common SIP and/or IAX registry. The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com

Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (=> one single IP with different SIP ports), the last entry into my

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Hi All, I have 2 SIP clients configured and connected to Asterisk. When I place a call from SIP1 to SIP2, if both codecs are the same then everything works as expected. I then allowed one of the clients to use alaw instead of ulaw and there were audio problems (couldn't hear the other end, etc). Same thing happened when I tried to use gsm<->alaw/ulaw. Any ideas? I'm using

Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 <---RTP---> Asterisk <---RTP---> SIP2 I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle

Hi thanks to everybody who has been assisting me in solving the various problems I had to dial out from Asterisk to a PSTN number with SIP using Nikotel's VoIP service. I have drafted a mini-how-to which is available at http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf This is a first draft, I will amend this further, in particular the "verify and debug" section

I'm trying to use accountcodes, but experiencing inconsistant results. I have two * servers, one which appears to be working as expected and one not. I would like to prepend the device's accountcode to the dialed number. The sip1 server does not seem to have the ${ACCOUNTCODE} variable set when reading the extensions.conf, but sip2 server does. What troubleshooting or trace

Here is my scenario: Sip phone number 1 and 2 are defined in sip.conf, and both have musiconhold=<class> set to the same outbound class that I want. This works fine for outbound calls (out to the pstn) Also, in extensions.conf for each extension that is setup to dial each of those sip phones, the first priority is SetMusicOnHold(<class>) So this works when a call comes in from the

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten =>

Hi List, I have a Multitech H323 Gateway MVP400 box with 1 phone on port FXO 1. I have Asterisk ruuning in Fedora Core 3. Both are in the same network. But I can't figure what I have to do in Asterisk to make that box work. What files I have to configure? Can anyone help me? I will really appreaciate youu help. Luis. _________________________________________________________ Do You Yahoo!?