search for: ougoing

I'm able to receive but not make calls with zaptel using an X101P connecting to Asterisk with an Xlite client. My client has context = flat in sip.conf and extensions number 8919 In extensions.conf I've got: [home] ; Line 1 ; exten => 8919,1,Dial(${PHONES1},20,Ttm) exten => 8919,2,Macro(vmessage,${PHONES1VM}) exten => 8919,3,Hangup [outgoing] exten =>

Hi list, I would like to know of a possible way to dial a pstn number with an extension . Let the number is 56626965-234 so now i wanna dial 56636965 then wait for some time and dial the extension 234 to reach a particular person.I am afraid that i could not figure it out. I am trying in this way.. [outgoing] exten=>_566X.,1,wait,2 exten=>_566X.,2,Dial(${EXTEN})

...e websites from my CentOS 5 box Target websites: www.connecttech.com www.3ware.com (two of my HW vendors) I can usually get some kind of response, but if the content (download or page itself) is larger in size (downloads never pass 100K), then it hangs... When I fire-up wireshark, I get a lot of ougoing highlighted Checksum Errored packets but I don't know what's causing it... Here's what I've eliminated thus far: - Websites are up and responsive - Cable(s) is fine - Network drop is fine - I put myself behind a firewall to make sure it wasn't my network - Network card has bee...

...al IP. The -j SNAT --to-source ot -j MASQUERADE dont work, are ignored, and I dont see any packet through these rules in iptables -tnat -L POSTROUTING. I used tcpdump to trace packet on the physical server on virbr0 interface and on eth0 interface. I see the packets on outgoing route. But, the ougoing packets are presented to the external interface with the internal address 10.0.0.x instead of the address specified in the -j SNAT rule. Am I the only one in this case? Somebody could help? Thanks Patrick

Hi, I'm experiencing some voice delay (2-3 sec) after outgoing call is setup. It means during the first 2-3 secs, audio is very choppy or nothing. So usually I can't hear the 'Hello". I use IAX2 for my ougoing calls with Grandstream phone as a client. Any hints to prevent this? Thanks, David

OK, so I answered my own question. Turns out case #2 just goes to extension 2. Still trying to figure out the optimum arrangement so I don't have an inordinate number of extensions. Maybe like this: 1. First outgoing try 2. Second outgoing try 3. Third ougoing try 4. Play a message and/or hangup 102. Goto 2 203. Goto 3 304. Goto 4 >> But this is not to say _you_ can't built a reliable VOIP based >> system. Get _two_ providers and set up your dial plan in >> extensions.conf to "fail over" if one service fails to >&gt...

...ks a one second?audio file. There will be no "long coversations" or any nonsense like that. The Manager API is working great, but it seems to "queue" the outgoing calls and only does them one at a time through my VOIP provider (teliax.com). ? 1) Is the max number of simultaneous ougoing calls solely dicated by the VOIP provider, or is it limited by * ? ? 2) Does IAX2 have an advantage over SIP in this regard? ? 3) I would like to have about 50-100 simulatneous outgoing calls, what do I need to do to accomplish this? ? Thx, Tom ? ? Do you Yahoo!? Read only the mail you want - Yaho...

Hi, I'am new to * and googled/read a lot, but did not find (yet) a lot of info to do the above. Some months ago, I did find a 'story' from somebody having put * between his PRI and current PBX as IVR, but I can not find it back :-( Any help/pointers are appriciated. Txs alex -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs

Hello, I searched the archives mailing list of LARTC. Everyone discussed about marking outbound ftp traffic . I could not find any thread discussed about marking inbound ftp traffic. With inbound ftp traffic , we don''t know the random ports specified by ftp servers in passive mode ? So marking inbound ftp traffic is impossible ? If it is possible, can you tell me, Thanks in

Perfect, thanks very much hth. I just set it to unknown, but it doesn't work. Have I to use also prilocaldialplan ? Thanks again Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson Inviato: gioved? 29 settembre 2005 16.22 A: 'Asterisk Users Mailing List -

Hello, I have som problems with my BRI/capi setup. I manage to call in to the system (some rows below). ---------------- -- Executing Dial("CAPI[contr1/16453]", "SIP/BYEXTENSION@janm|10") in new stack -- Called s@janm -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing ---------------- But I can't make outgoing calls from

Hi, we are using asterisk with Junghanns QuadBri and some sip phones. 2 channels are configured in NT mode (ISDN PBX connected, internal ) and 2 channels are connected to the public ISDN network (bri-cpe). We use Bristuff 0.2.0 RC8C from Junghanns. When a call comes in from the public phone for a specific extension (Hotline Number), we initiate a parallelcall to some SIP phones and also to our

I am setting up my * for at home office and still have analog phones attached and answer from those analog phones and not necessarily through the pbx. I found that with the X100P cards, they see the 2nd ring and will be ready to answer the line. I used a Wait to pause and allow another 2 rings before * answers. But found that if we answer the line after the 2nd ring and before the 4th, * still

Hello all, I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a Digium 8 port FXO card. The local network is 100Mbps Ethernet and my phones are Linksys SPA-921 or Linksys Analog adaptors. The phones are setup with DHCP, and are on the same flat non-routed network. There is no NAT involved. If I call from extension 6000 to extension 6001, or vice-versa both are SPA-921s.

Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>