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Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at

What is the consensus on using the 1.4 jitterbuffer? Do most people enable it? We have a "PSTN" server that has our RBS T1 trunks in a central location, then have clients that connect via SIP to us for access to those trunks. Most of them are just fine, but lately we have a handful that are having latency and jitter issues. I am hesitant to just turn on the jitter buffer in

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

Hi List, Asterisk 1.4.22 built by root @ carl on a i686 Purely SIP Linksys SPA962 with 932 sidecar and also Cisco SPA508 / 525G with Sidecars Have an issue with this happening with a number of my customers. Customer hits the ringing BLF on the sidecar to pickup the call incoming on another handset. Not always but enough to cause me a problem it leaves a channel open. At the moment, this is

Hi All, Is it possible to specify more than 1 localnet? I know this is an odd question. I have a customer that has multiple sites linked by VPN. Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24 We want to allow some access to the public IP address at the main site. For this to work I need to use the externip and localnet directive. If I do this it rewrites the SDP with the

Hi List, I need to modify the callerID name of the call coming back when a parked call returns to the extension that parked it when it times out. Looking at app_parkandannounce.c /* Now place the call to the extention */ snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars =

...an intercom/paging system > (Tech Support) > 7. Asterisk on OpenWrt (first time user) (Sebastian Kemper) > 8. Dahdi ISDN logging (Grant Bagdasarian) > 9. Re: Dahdi ISDN logging (Tony Mountifield) > 10. UNREACHABLE peer (thufir) > 11. Re: UNREACHABLE peer (dotnetdub) > 12. Re: UNREACHABLE peer (thufir) > 13. Re: UNREACHABLE peer (thufir) > 14. Re: UNREACHABLE peer (thufir) > 15. Re: Caller ID Names (Jordan Cook - Gyron Networks) > 16. Re: Caller ID Names (Jordan Cook - Gyron Networks) > > > -------------------------------...

Turn on sip debugging for this peer and watch for the options sending and response. If you are getting a response to your options asterisk shouldn't be marking the peer as unavailable. is your asterisk behind a firewall? On 20 March 2015 at 13:42, thufir <hawat.thufir at gmail.com> wrote: > I wasn't able to get much out of babytel, beyond the fact that I was, > apparently,

Hi List, Anybody any ideas on these? [Nov 26 15:14:10] WARNING[3265] chan_sip.c: Remote host can't match request NOTIFY to call '1c4890a52552c39b0b81702353087bf0 at 192.168.33.12'. Giving up. [Nov 26 15:16:44] WARNING[3265] chan_sip.c: Remote host can't match request NOTIFY to call '7920154238a73e5643a899635b0cb601 at 192.168.33.12'. Giving up. [Nov 26 15:23:37]