similar to: videosupport = yes -- how to use it?

Hi All First time poster, long time reader. I just ran into something really bizarre. I've enabled videosupport and have been testing sip calls with Xten Eyebeam software, it works (mostly) However, when I have Videosupport=yes In the [general] section of my sip.conf, broadvoice calls fail w/ "We're sorry your call cannot be completed at this time" So... I've

Hi all I was tryed to connect to mysip.ch scs_client by siemens that isn't works well. Does anyones knows to work H/W or S/W applictations in asterisk SIP videosupport? Regards mack_jpn

Dear all, I've implemented an Asterisk 1.4 with SIP service for voip and video. So I can establish a voip + video connection *one-to-one* only....it works OK. But I'd like to implement a videoconference *one-to-many* in order to intercommunicate many clients, is it possible with Asterisk 1.4 ??? (multicast is better than brodcast in this situation of course) Thanks a lot, Alejandro

Hi, I have a samba 4.6 member server joined to an 4.6 AD. There is a share named [videosupport2] with access rights for a user Videosupport. Now I’m able to connect to the share from an macOS client by share name videosupport2 AND just videosupport without the „2“ at the end …. Any hints why? Regards . Götz

sorry, no webchat, is a videoconference program. And I'm using UDP as you can read down on the mail. The problem is that speex packages alone work great, but to split speex/theora packages on the arrive I need to use some kind of container, and I'm trying ogg. On 8/7/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > If by webchat you mean anything interactive,

I am a fairly novice Asterisk 1.4 user who used to use CallWeaver, based on asterisk 1.2. I used Realtime MySQL with CallWeaver and am currently using the very same MYSQL tables (and columns) with Asterisk 1.4.11 and things are working well. The questions I have are, since new configuration variables have been added into Asterisk 1.4, can I simply add columns in my MySQL sippeers table for

Hi all, A Cisco ATA186 configured with g711ulaw, NAT=yes and canreinvite=yes, made calls through Asterisk to a Cisco 5300 gateway out to a PSTN line with errors as follows: -- Executing Dial("SIP/ata186-c1cf", "SIP/29086988@110.100.231.2:5060|30|r") in new stack -- Called 29086988@110.100.231.2:5060 Apr 15 16:11:22 WARNING[1116941120]: chan_sip.c:2049 process_sdp:

Hi all, for some strange reason, our still-under-test Asterisk deployment wants to contact the outside world and that raised some eyebrows here... Just a sample of our firewall log: -- ...a=DROPIN=eth0 OUT=eth2 SRC=192.168.36.199 DST=195.77.113.194 LEN=476 TOS=0x10 PREC=0x00 TTL=62 ID=39572 DF PROTO=UDP SPT=5060 DPT=62975 LEN=456 -- Why is this happening? We got no relationship with the DST

Hi, Anyone here using sipgate.de ? It worked for months, but for a couple of days now I'm unable to register with them. My account is ok, because I can login to the website. Asterisk keeps showing me: Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n) I looked at the sip debug stuff, and all I

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

Hi list, surely this has been posted before but the archives don't offer a 'search' functionality and I need an answer really soon on this subject... so, my apologies. Which ports (range) must be open on a firewall, either TCP and/or UDP, for Asterisk to work correctly? TIA, Martin

Hi there, the subject says it all... I get the following error messages: ---- mmielke@pbx:~/chan_capi-0.3.5> make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '0426000000' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension

Hi all, I compiled/installed chan_capi.so without problems. When I launch Asterisk, I get the following error: --- [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif: ast_capi_pvt(91xxxxxx,*,pstn,0x2,2) (1,2,64) (0)(0.800000/0.800000) 0 Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338

Greetings, I am attempting to configure an AS5300 to provide a SIP based gateway to the PSTN from Asterisk. I have been unable to identify through the docs how specifically this should be configured in Asterisk and have not been able to get things working through trial and error. I am sure I am missing something fairly obvious here but any guidance (or example cfgs) would be much appreciated.

Guys. I need some advice on some h323 issues. I need to test connectivity from Asterisk to a Cisco AS5300 that has PSTN lines and to cisco 3600 voip routers. H323 needs to be used here but I was wondering if anybody has linked Asterisk to these Cisco routers before? Thank you for any pointers.

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

How can I get rsync to write a log record (like --log-format does for data transfer) when no content needs to be transferred but attributes (owner, group, mode, ...) are changed? -- Dave Mielke | 2213 Fox Crescent | I believe that the Bible is the Phone: 1-613-726-0014 | Ottawa, Ontario | Word of God. Please contact me EMail: dave@mielke.cc | Canada K2A 1H7 | if you're

Hi guys, I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC, with eth0 set to address 192.168.1.1 (NATted over public network, with address 89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) Then I have configured an account as following: [999] type=friend