similar to: Asterisk and grandstream weird call probs

Hey all, I've been trying to get supervised transfers working without success. I'm currently running 1.0.7-stable and think it might be a version problem. Is the supervised transfer feature available in 1.0.7 or do i need to suck down a new version from CVS? Otherwise, apart from setting up features.conf, is there anything else i'm missing? TIA, Jamie. -- Jamie Carl

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

Hello, Has anyone gotten a Grandstream BT100 to work with Level 3's 3Tone? I've been able to get my extension to interface with it, but there is no sound and the call on the GS side terminates prematurely. Here is the relavent portion of the SIP.CONF [4007] ; Budgetone BT100 type=friend insecure=yes context=test-budget username=4007 fromuser=4007 callerid=4007 host=dynamic nat=yes

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't

Hi there, for those that haven't yet found out for themselves: - BudgeTone/ HandyTone firmware now has an option for "disable callwaiting" which probably eliminates the most urgent need for outgoinglimit= and incominglimit= in sip.conf (firmware 1.0.4.54 and later, maybe even available in some slightly earlier versions) - new option "subscribe for MWMI" (message

dear all, can anyone assist on how to enable callerID to appear on grandstream bt-100 sip phone. [1003] type=friend context=default username=1003 secret=**** ;fromuser=1003 ;callerid=John Doe <1234> host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 incominglimit=1 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default" disallow=all allow=ulaw allow=alaw

Hey all, I've been trying to get my X101P working again as of late (it used to work great) and before I decide to trash the card I thought I'd post up my symptoms to see if anyone has any ideas. My old working config was basically 1 channel running fxsks signalling. It was working great with no echo, busy detect worked well and I was very impressed considering this is all off and

I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? When people use asterisk on a broadband connection used

Hi All, This is the first time I'm submitting a patch, and I hope it fixes more than it breaks. I'm putting it here, since John Todd mentioned a while ago about the heavy load Mark and crew have at Digium (doing such good work), so I thought all of us could test this first, and if ok submit for inclusion in CVS later if appropriate. This is an extension to work done earlier (sorry I

Hi all, I have some peers configured in SIP.CONF file with parameters incominglimit and outgoinglimit set up to 10. By doing that, I expect that this peer will not be allowed to handle more than 10 incoming calls and 10 outgoing calls at the same time. However, since I upgraded to Asterisk 1.2.17, I started to face a problem. Sometimes, calls to those peers are not connected. When I check the

Hi, all. I have problems with iax2, when try to communicate with one third server, asterisk reports the following errors in server's, could help me? Server A it speaks It with C in iax and Server B it speaks with D in iax, but Server A it does not obtain to speak with B in iax, reports the following error in server B "chan_iax2.c:5749 socket_read: Host 200.xxx.xxx.xxx failed you

For those that don't read every line of source code here's something I found out today... -------- Deprecated incominglimit and outgoinglimit Incominglimit = number of calls the local extension can originate to Asterisk. Outgoinglimit = number of calls Asterisk will terminate to local extension. End of Life for these commands announced**, please use setgroup and checkgroup, that will

Playing with hints/presence/BLF on asterisk I've made the following "discoveries". 1. The wiki at http://www.voip-info.org/wiki/view/Asterisk+presence says: "If you add incominglimit=1 to your peer in sip.conf, the SIP channel will notify you when that extension is busy." As "incominglimit" is obsolete you can use "call-limit".

The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing

Hi, This is a message I already posted on the chan_sccp mailing list, but since this list has a lot of active members, I'm hoping someone might be able to help (And my problem is * related, so I guess it's ok if I post it here also ;) ). I'm trying to get SCCP ATA188s to run with Asterisk. The Asterisk box uses the latest Asterisk@Home image (Version 2.6). I have compiled and

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

I have 2 issues which I need to resolve on our production Asterisk server: We are currently using Polycom IP600 VOIP phones for our office which are capable of handling 2 calls per SIP registration. What we're finding is when staff are on the phone, Asterisk will pass them a second call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I

Hello, Right now I have app queue working with incominglimit=1, there is no call waiting signal, but after a while( like couple of hours) some phones randomly get stuck. The * thinks that they are in use and doesnt ring them, when they are infact not in use. sip show inuse, shows that they are inuse. typing reload on the console resets this and they are again available for working. anybody

Not being an asterisk expert, but having been around the block once or twice when it comes to data and the like, I have made some observations based on the examples given on voip-info.org Sip configs. it appears there is an adjustment to be made in the sip_buddies example table: >>> name Although set to 30 characters, I don't see where it is limited in the text file. In theory,

Good morning How I use the described commands below to limit the number of simultaneous calls saw voip providers that they can be effected and be received in the trunk in the Freepbx? I verified the commands incominglimit and call-limit as I can use asterisk is version 1.4! It would like to restrict for I number it to four of calls that can be used in one trunk of a voip provider? thanks.