similar to: setmusiconhold doesn't work between 2 SIP phones

On Thu, 2006-03-02 at 11:42 -0600, Jordan Novak wrote: > Does anyone have a way to do wake calls? > > > > Jordan Novak > > Communications Technician > > Logistics Health Inc. You could use cron and /var/spool/asterisk/outgoing scripts to dial numbers, etc... > Can you elaborate, I am fairly new to Linux and a phone guy to boot. I am looking for a way for the

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

Hello to all. I have a couple of budgetones connected to Asterisk server. I can establish calls using budgetone with no problem, but when I hang up a Budgetone, Asterisk does not detect the hangup. It seems that the communication goes on in spite of budgetone's hangup. My sip.conf: [general] disallow=all allow=ulaw bindaddr=172.16.60.21 [sip1] callgroup=1 pickupgroup=1 type=friend

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

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

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

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 --

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

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

I get the following message on a Centos 5 system (really a Trixbox 2.4 build on Centos 5): Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port 1 disabled What does this mean? This message occurs about 30 times/sec for about 45 sec. Then my Bluetooth token starts up. Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port 1 disabled Jan 14 00:13:00 sip2

I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe

Hello, I'm trying to do something really easy : transfer a PSTN call to a H323 client. This works great. Now I'm trying to use the SetMusicOnHold function. I din't find any doc about it, I've just seen some mails in the list archive, but it still doesn't work. That's my extension.conf : [incoming] exten => s,1,SetMusicOnHold,default exten =>

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, 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

Hello All, Is the application SetMusicOnHold always followed by the Dial application? I'm studdying the info on Google and Wikki about SetMusicOnHold, and I'm not sure I understand what I found. Thanks, Anon

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

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

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

Gentlemen, Forgive me if I am posting at the wrong place! I was going to test the "new" chan_ooh323 driver so I did install: debian: Linux sip2 2.6.26-2-686 #1 SMP dahdi-linux-complete-2.2.0.2+2.2.0 Asterisk SVN-trunk-r231692 Did enable chan_ooh323, everything compiled without any problems. Hardware setup: Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975) X-Lite can