similar to: Blind transfers security

Hi, I've got a problem with rtp keepalives. I'm using basically the same config on 2 hosts, but one of them sends rtp comfort noise when it's on hold, the other doesn't. The only difference I can think of now is that one of the machines is multihomed, but that might be unrelated. rtpkeepalive is set to 2 and I can confirm is by doing `sip show settings`. I've tried all

Hi, I'm having problems with sip remote attended transfer using 2 asterisk boxes (same version, latest 1.4.X). Whenever I transfer from a call from box A to a call on box B, one call leg of the transferring phone is not disconnected (the one that is normally dropped by server side, phone disconnects the other one). The same situation works perfectly with local attended transfer. Is anyone

Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on

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Hi, I'm trying to access audionativeformat / other codec variables in the hangup handler of a call (with ${CHANNEL(audioreadformat)}), but I get no response. Also 'core show channel ...' doesn't list those variables. Are they always set by asterisk, or only in some scenarios? It's a simple SIP-SIP call with audio passing through asterisk, same codecs on both sides. I see that

Hi Is it safe to use IMAP and POP3 simultaneously to access the same inbox (using Maildir structure)? Thanks! Stanis?aw -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 833 bytes Desc: OpenPGP digital signature URL: <https://dovecot.org/pipermail/dovecot/attachments/20180228/c8f47500/attachment.sig>

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

Destroy ACL-s on destination when no ACL-s differens between source and destination. Bug is somehow related with function send_file_name() called with negative file descriptor f. There is no such bug in 2.6.9 version, but there options "-X -A --deleted" can't be used (we have "Internal error: wrong write used in receiver."). If I fix this, avoid calling send_acl() &

I'm getting this error message and I don't really understand what rsync is trying to tell me:- rsync: link_stat "/rdiffBackup/gradwell/Mail/." failed: No such file or directory (2) rsync error: some files could not be transferred (code 23) at main.c(977) [sender=2.6.9] Can anyone explain what it's saying please. /rdiffBackup/gradwell/Mail/ does exist and is

Does anyone here use either Gradewell or inWeb for service? They are both UK based. I'm trying to get a couple of inbound IAX2 based numbers from both of them to work with no luck at all. The one thing that sets these guys apart from the rest of companies offering inbound numbers is they tie the account to the IP of the asterisk server. Neither use register lines in iax.conf, there appears

Hi All, I'm looking for how to enable SIP Markers, or specifically, how to have the TIME reset when a call route changes. I'm debugging an issue, where a sip client we have switching to one-way-audio, when an asterisk server fruther down the call path dials out to the PSTN. Scenario is: SIP Client -> A*k1 -> A*k2 -> PSTN Provider/Gradwell -> O2 -> Mobile

Hi, I'm using a loop to extract 2 columns from an array into multiple files. I can use the following to export 3 files, containing column 'ID' with one of the three event columns. > ID<-c("A","B","C","D","E","F") > event1<-c(0,1,0,0,1,0) > event2<-c(1,1,0,1,0,0) > event3<-c(1,0,1,0,1,0) >

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

All, I've been experimenting with shortcodes for SIPgate etc. Passing calls to SIPbroker seems a good way to go, but the message I've had back from SIPgate is "we don't support SIPBroker"... So whats the easiest way to support SIP <> SIP network calling? At the moment, I've setup some local shortcodes (eg dial **777. to goto sipgate.co.uk) based on what Gradwell

Hi, I realise this is probably the wrong list for such a question, but I need a pointer to somewhere I can get some feedback on experience of (business class) voip providers for the UK? Situation is that we are currently with Gradwell and use them for an inbound/outbound single line for a business and their quality has gone from excellent to abysmal in the last few weeks. I'm sure they

Mar 29 10:57:02 k8ux dovecot: POP3(stf): mbox sync: Expunged message reappeared in mailbox /var/mail/stf (UID 123 < 60016, seq=2, idx_msgs=0) Mar 29 10:57:02 k8ux dovecot: pop3-login: Login: user=<stf>, method=... Mar 29 10:57:03 k8ux dovecot: POP3(stf): mbox sync: UID inserted in the middle of mailbox /var/mail/stf (60016 > 123, seq=2, idx_msgs=1) Mar 29 10:57:03 k8ux dovecot:

Dear R Users, I run simulation that takes very long time (R 2.2.1, Win XP pro., Rgui SDI mode, editor Tinn-R). It's happened that R shuts down and Windows display the message: Rgui.exe makes an error and the application will shut down. Unfortunately everything I lost. Below I paste the message that is created when error appear. Maybe You as an expert will figure out what is happening to me.