Re: [asterisk-users] To Header instead of Request URI based routing

Hi, do you have access to the system that sends you these calls? If it's also an Asterisk, you could tell it to send another INVITE URI, regardless of what is submitted in the registration. On Asterisk with chan_sip you can do it by dialling:

Re: [asterisk-users] Need to restart Asterisk if remote server not working?

Hi Luca, Am 06.05.2017 um 15:49 schrieb Luca Bertoncello: > I'm running an own BIND on my Linux-PC... Me too ;-) > Maybe should I configure a forwarder for the zone t-online.de? It not > difficult, and if you mean it can help, I'll do that... In the meantime, I setup forwarding requests to

Re: [asterisk-users] Need to restart Asterisk if remote server not working?

Hello, I'm also a customer of the DTAG. Yesterday, the messed a bit with their DNS entries... If you are NOT using their DNS resolvers you got a "wrong" IP address back that was not working. Besides that, you should disable SRV lookups for their SIP peers. Since Asterisk's chan_sip.c does not

Re: [asterisk-users] How to have callers not being billed when in waiting queue ?

Hi, in Germany, this kind of regulation is in effect for phone numbers which cost more than a normal landline call. The regulation states, that the waiting time must not be charged to the customer. Most companies implemented this by simply switching their telephone numbers to those, which

Re: [asterisk-users] multiple outbound invites

ason anyone can think of that our asterisk (11.11.0) would suddenly start > doing this? It may be that it has been doing it all along, and our carrier > just started rejected calls that come in this way, I'm not sure. > > Cheers, > > j > > -- Viele Grüße aus dem Ta

Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

Am 16.02.2017 um 15:01 schrieb Joshua Colp: > As for your issues please do file them. I'd also suggest using bundled > PJSIP, it works the best with Asterisk and we backport applicable fixes > and include fixes we've created that have not yet made it to a PJSIP > release. OK, I'll try again

Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

Hi, Am 16.02.2017 um 14:19 schrieb Annus Fictus: > And Microsip using PJSIP SIP stack :) Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not sure if this really is a sign of good quality. Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP

Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

Hello, I'm a big fan of PhonerLite. It's more poplar in Germany, but also available in English language. This client supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm Yes, the GUI is not that much user friendly as Zoiper is - but at least a very good and stable client for

Re: [asterisk-users] SIP host name resolution

Hi, Am 03.02.2017 um 18:23 schrieb Steve Edwards: > If I have a SIP endpoint defined in sip.conf using a host name instead of an > IP address, do I have to reload sip to get Asterisk to 're-resolve' the host > name if I change the IP address in my DNS? Normally, Asterisk honours DNS TTL and

Re: [asterisk-users] Connection dropped after 15 minutes with Deutsche Telekom

Hi, I figured out that this happens, when Asterisk ignores Session-Timers requests. So I added the following to my DTAG peer configuration and eleminated the problem - and can use g722 on the DTAG network :-) --- session-timers=accept session-expires=120

Re: [asterisk-users] new inbound DID provider... no auth?

Hi, That's right - you just need to define a peer with a static IP address and "type=peer" to assign incoming calls to a peer name and apply the corresponding configuration (e.g. codecs). To make your configuration less redundant you can use templates in your peer definition (at least for

Re: [asterisk-users] Change Media IP in SDP

Hi, normally, Asterisk handles RTP IP addresses in SDP correctly, if you have specified - that NAT traversal is enabled for all peers (e.g. nat=force_rport,comedia) - your local network with "localnet=yournetwork/networkmask" - e.g. "localnet=192.168.1.0/255.255.255.0" - directmedia,

Re: [asterisk-users] Touch tone stutter

Hi, you could try switching the DTMF mode of the ATA's SIP peer (and also in the ATA itself) to INBAND transmission. In this mode, the ATA doesn't need to recognise DTMF tones and your Asterisk can interpret it. For this to work, the ATA needs to use a G.711 codec. Inband DTMF needs an

Re: [asterisk-users] Non-global variable that follows channel?

Hi, is channel variable inheritance working for your setup? Passing variables to other channels can normally simply be done by naming the variable with one or two prefixed undersorces to make it available to the channel that is created from that one defining the variable. But I have no idea if

Re: [asterisk-users] Asterisk 11.24.1 garbled audio

Hi, Am 17.11.2016 um 13:51 schrieb Jerry Geis: > PBX Core settings > - > Version: 11.24.1 > Build Options: LOADABLE_MODULES, BUILD_NATIVE > Maximum calls: Not set > Maximum open file handles: 1024 > Root console

Re: [asterisk-users] Asterisk 11.24.1 garbled audio

Hi, Am 15.11.2016 um 17:52 schrieb Olivier: > Hi, > > How can I double check which timer is currently is use in a running system ? > core show settings doesn't tell anything, if I'm not mistaken. To determine which timing module is currently in use, you can take a look at "module show like

Re: [asterisk-users] SIP and RTP port and IP addresses

Hi Ethy, Am 09.11.2016 um 17:13 schrieb Ethy H. Brito: > How are these parameters available from dialplan? > > For instance, ${SIPURI} holds the internal "IP:port" if the client is behind > NAT. > I need the external IP:port You can get the peer's signalling IP address from

Re: [asterisk-users] Suddenly getting lots of "Unable to send packet: Address Family mismatch between source/destination" but ONLY on 1 of 2 VPSs in same datacentre.

Hi Jonathan, Am 05.11.2016 um 14:08 schrieb Jonathan H: > What I don't understand is that while Ubuntu has IPv6 of course, the VPS host > is set to V6 disabled. and as far as I am aware, and my ITSP doesn't have > IPv6, so I just can't figure out why two IPv4 systems are getting IPv6 >

Re: [asterisk-users] Just got defrauded - how do I block calls which contain a dash (RegEx noob question)

Hi, Am 28.10.2016 um 17:38 schrieb Markus: > exten => _-.,1,NoOp(Blocking dash) > exten => _-.,n,Hangup > How do I do it right? why not using FILTER() in your dialplan to eleminate all chars that are not numeric? Like Set(VAR=${FILTER(0-9+),${EXTEN}}) That would eleminate all

Re: [asterisk-users] Adding a pause when transfering a call

Hi, some phones can add a pause when dialing, sometimes by holding the * or # key a few seconds after the first digit. If it works, the phone normally adds a "W" or ";" to the dial string. So you would program the speed dial key with <*2[hold * or #]101>. Am 01.10.2016 um 20:22 schrieb Tech

Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

Hi, OK, then it looks like the client transferred the call anywhere else. Do you see an entry in your log that refers to the bridge ID 00bd58c3-3bce-4f1b-9d79-11eb96f37260 ? If there was a transfer, the call *may* have been bridged with the transfer destination. Also, the destination might be

Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

Maybe the client just put the call on hold. So the call technically has not ended AND the client does not need to send or handle any RTP data. Is there any mention of "music on hold" for this channel? Greetings Max - Nachricht von Leandro Dardini - Datum:

[asterisk-users] Get Realtime extension matched entry ID

Hello, is there a possibility to get (by dialplan variable?) the entry ID of the realtime extensions table, that matched the current call? For example (simplified): ID - exten - 1+49123456 2_+49555. If I receive a call on +49123456 this surely

Re: [asterisk-users] Asterisk 13.11 realtime problem registering phones

Hi, Am 02.09.2016 um 22:48 schrieb Carlos Chavez: > I upgraded my office installation from 13.10 to 13.11 yesterday and now I > am having problems registering phones. Here is what I get on the CLI: > > [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: > Realtime

Re: [asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily

Hi Jonas, Am 02.09.2016 um 11:26 schrieb Jonas Kellens: > [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from > 11.22.33.44:40670 > [Aug 31 14:59:34] -- Now forwarding > Local/myaccount184@CallFromQueue-07f4;2 to 'Local/23@from-internal' > (thanks to