similar to: Call Transfer question

On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad <faheem2084 at gmail.com> wrote: > > > On Wednesday, 14 September 2016, Madushan Geethanga < > mgliyanage.rc at gmail.com> wrote: > >> Hi, >> >> What is the equal option for externip in asterisk 13 with pjsip. I have >> tried >> >> external_media_address=XX.XX.XX.XX >>

Well, I thought I had the problem solved. Ported everything over to PJSip and build RDNS records for the phones and the server, but I am still experiencing the problem on incoming calls. ** On 6/7/2016 1:00 PM, Faheem Muhammad wrote: > I've faced the same issue. The issue was related to DNS, the reverse > lookup query failure caused the delay around(7-9 seconds). The purpose >

Hi, I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan

On Thu, Sep 15, 2016 at 8:38 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Thanks for the reply. > > Yes my PABX is on the cloud when I try to register to the server, the > server sends registration OK with public address but OPTION method > includes the private address of the server in from header not the public > address. I have include

Hi, I'm getting an issue while executing AMI Originate. I'm getting "extension does not exists" on Originate's Response, and on the other hand Asterisk CLI say "fwrite() returned error: Broken pipe" Please suggest me what is wrong. Muhammad Faheem ### my originate code block ...

On Fri, Sep 16, 2016 at 5:55 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Tried with both softphone (Ekiga) and snom IP phone, contact header > contains the public IP. but from header contains the private IP. after > OPTIONS method sent by the server. client sends an Register with expires 0. > Ok, did setting from_domain work? > > Best

I have asterisk 13.5 configured with a simple dial plan, 3 SIP clients two Laptops and smartphone with softphones installed. Now I am trying to store cdr into a database but not able to make a connection of ODBC drivers to MySQL is there an option or anything. Thanks in advance My configuration:: *sip.conf* [general] trasport=udp ;Data format | sample commennt [template01](!) type=friend

Hi, I'm getting errors while originating a call through AMI. [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe Asterisk version 11.0.1

Hello, Let say all the SIP devices will be registered on the proxy like kamailio. Agent is a member of Support and Billings Queues on the asterisk servers. Support queue on "Server A" and Billings Queue on "Server B" for example. This will be done via RealTime Queue. I want Agent to dial 1234 on a sip device and it will prompt to enter a pin number to Login via

Hi all Is there anyway i could get in the dialplan the amount of time a caller waited in the queue before exiting? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160518/b3b082aa/attachment.html>

Hey,?is there any Diguim distributor in Lahore,Pakistan? I need to buy X100P.? Muhammad Faheem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100326/104853f8/attachment.htm

On 2016-08-09 10:06, Faheem Muhammad wrote: > trip time and Call Setup time of SIP Requests. > In case of GSM Network with high delay you need to set the T1 timer a > higher value like 1000ms (500 ms default). Similarly you can reduce the > Call setup time by configuring 'T2' upto you choice as per you telephony > network. Configure t1min, timert1 and timerb according to

Hi, What is the equal option for externip in asterisk 13 with pjsip. I have tried external_media_address=XX.XX.XX.XX external_signaling_address=XX.XX.XX.XX but asterisk 13 writes local ip to the from header. any suggestions? Best Regards, Madushan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I try to find informations concerning Mixmonitor command, but ... without success. MixMonitor command take at last parameter "command". This command can be a shell script. When record is over, and this command executed, asterisk wait for a return code or asterisk move to the next dialplan instruction ? This command is a background task or use ressources in asterisk ? For

<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p>Hello everybody,<br><br>I manage not to detect one silence with record () when I make as follows:<br><br>Exten = > 0178900271, n, Record ($ ${ link_recorded_pseudos_clients } pseudo_ Client_Id} wav, 5,5) exten = > 0178900271, n, GotoIf ($ ["

Hi, I'm running the following command as a local copy command. faheem at bulldog:/mnt/data$ sudo rsync -abvz --super /data/ . Origin directory faheem at bulldog:/data$ ls -la total 28 drwxr-xr-x 7 root root 4096 Jun 26 08:34 . drwxr-xr-x 25 root root 4096 Apr 13 17:09 .. drwxr-xr-x 2 owzar001 root 4096 Nov 6 2010 CTS drwxr-xr-x 2 owzar001 owzar001 4096 Aug 27 2010

Hi, We have been migrating our PBX system from Asterisk 1.8 and chan_sip to Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have stumbled on a behaviour difference I don't like. With chan_pjsip when a phone went unexpectedly offline (Ethernet cable disconnected) Asterisk would detect this quickly (through the 'qualify' pings), mark the phone as 'Unavailable' and

Hi, I'm seeing something that may be a bug in R's standalone math library, which is packaged by Debian as r-mathlib. I reported it to the Debian BTS as http://bugs.debian.org/657573 I'm using Debian squeeze, and the code was tested with r-mathlib 2.11.1-6 (default on stable) and 2.14.1-1 (from testing/unstable). I summarize this report below. The following code with the R math

Dear rsync people, I have just started using rsync for backups. I have had a couple of issues. Note I'm trying to use rsync as user using ssh between two machines both running Debian GNU/Linux potato (2.2r3). The local machine is currently running 2.4.6-1 and the remote 2.3.2-1.2. 1) When I run rsync with the vv option, stuff scrolls of my screen faster than I can read it. I was wondering if

Dear People, I want to run gcc with optimisation turned on (-O2), and with -Wall (all warnings) enabled, when using R CMD SHLIB. When I do make, which is R CMD SHLIB -Wall -O2 cftp.c mcmc.c latticefn.c -lm in this case, I get faheem ~/research/cftp>make R CMD SHLIB -Wall -O2 cftp.c mcmc.c latticefn.c -lm make[1]: Entering directory `/home/faheem/research/cftp' gcc -I/usr/lib/R/include