Re: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450
Thank you all, After setting the phone to factory defaults, entered configuration parameters, phone is working again. I really don't know why all sudden stop working. at least know i have a working phone I will go thoroughly through the logs, I hope to find the answer, if I do I will post it here. Thank you again. On Thu, Jan 2, 2014 at 9:12 AM, Ryan Wagoner wrote: > > On Thu, Jan 2, 2014 at 11:13 AM, Eric Wieling wrote: > >> Which firmware version? 4.1.x is only for use with MS Link server. A >> symptom of running 4.1.x firmware with a non-MS server is the phone will >> not show buddies. >> >> > I'm running 4.1.0 on a Polycom IP 335 and IP 550 and version 4.1.5 on a > Polycom VVX 400. Buddies work on all three phones. The firmware is for both > SIP and Lync. You change the base profile option accordingly. Look in the > Polycom UC Software Admin Guide for more information. > > Ryan > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450
On Thu, Jan 2, 2014 at 11:13 AM, Eric Wieling wrote: > Which firmware version? 4.1.x is only for use with MS Link server. A > symptom of running 4.1.x firmware with a non-MS server is the phone will > not show buddies. > > I'm running 4.1.0 on a Polycom IP 335 and IP 550 and version 4.1.5 on a Polycom VVX 400. Buddies work on all three phones. The firmware is for both SIP and Lync. You change the base profile option accordingly. Look in the Polycom UC Software Admin Guide for more information. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk
Adam, Thanks, I will try that this afternoon. JohnM On 01/02/2014 11:31 AM, Adam Moffett wrote: > top posting is superior anyway --- *ducking to avoid thrown objects* > > If I recall correctly, when doing something like that with a polycom I > had to set the registration interval absurdly low, like 20 seconds or > something. I think the Polycom didn't send keepalives and that was the > workaround. > > >> top posting so as to not make thread even more confusing. >> >> Nick, >> I have nat=force_rport,comedia in sip.conf. It is my understanding that >> nat=yes is deprecated? >> >> Thanks, >> JohnM >> >> >> On 01/02/2014 10:51 AM, Nick Olsen wrote: >>> Make sure you have nat=yes in your sip.conf either under globals or >>> individual sip peer settings. >>> >>> Nick Olsen >>> Network Operations >>> (855) FLSPEED x106 >>> >>> >>> >>> >>> *From*: "John Millican" >>> *Sent*: Thursday, January 02, 2014 10:50 AM >>> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" >>> >>> *Subject*: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> >>> NAT/Firewall-> Asterisk >>> >>> Hello, >>> CentOS 6.x and Asterisk 11.x >>> I have an interesting, to me at least, situation. Using a Polycom >>> 501(also tried with X-Lite). I have set up Asterisk to accept >>> registration from the Polycom and it registers successfully but then >>> withing 30 seconds on the CLI I get the message that the Polycom is >>> unreachable. The phone still shows that it is registered and if I try >>> to place a call from the phone to my Cell, my cell rings once and then >>> stops. I get a packet retransmission error: >>> WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout >>> reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical >>> Response) >>> Followed by: >>> n_sip.c:4203 retrans_pkt: Hanging up call xx@192.168.0.100 - no >>> reply to our critical packet >>> I am "assuming" that there is a problem with NAT. I have externip set >>> in sip.conf. >>> Any pointers to what I am missing? >>> Thanks, >>> JohnM >>> >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >> > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk
top posting is superior anyway --- *ducking to avoid thrown objects* If I recall correctly, when doing something like that with a polycom I had to set the registration interval absurdly low, like 20 seconds or something. I think the Polycom didn't send keepalives and that was the workaround. top posting so as to not make thread even more confusing. Nick, I have nat=force_rport,comedia in sip.conf. It is my understanding that nat=yes is deprecated? Thanks, JohnM On 01/02/2014 10:51 AM, Nick Olsen wrote: Make sure you have nat=yes in your sip.conf either under globals or individual sip peer settings. Nick Olsen Network Operations (855) FLSPEED x106 *From*: "John Millican" *Sent*: Thursday, January 02, 2014 10:50 AM *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" *Subject*: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk Hello, CentOS 6.x and Asterisk 11.x I have an interesting, to me at least, situation. Using a Polycom 501(also tried with X-Lite). I have set up Asterisk to accept registration from the Polycom and it registers successfully but then withing 30 seconds on the CLI I get the message that the Polycom is unreachable. The phone still shows that it is registered and if I try to place a call from the phone to my Cell, my cell rings once and then stops. I get a packet retransmission error: WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical Response) Followed by: n_sip.c:4203 retrans_pkt: Hanging up call xx@192.168.0.100 - no reply to our critical packet I am "assuming" that there is a problem with NAT. I have externip set in sip.conf. Any pointers to what I am missing? Thanks, JohnM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450
Thank you all for your prompt reply, my phone was working up until this morning it just stop subscribing to the Asterisk server. Version: 3.2.4.0244 phone is configure to download configuration via ftp, again it configure right because it was working fine. the phone icon next to the extension number is dark same as the background so that means is not subscribing to the Asterisk server. Thank you very much. On Thu, Jan 2, 2014 at 8:19 AM, Kevin Larsen < kevin.lar...@pioneerballoon.com> wrote: > asterisk-users-boun...@lists.digium.com wrote on 01/02/2014 10:03:19 AM: > > > From: motty cruz > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > , > > Date: 01/02/2014 10:02 AM > > Subject: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450 > > Sent by: asterisk-users-boun...@lists.digium.com > > > > Hello, I'm having issues with my phone Polycom sp450 not subscribing > > to Asterisk server. Asterisk server is fine, firewall is not the > > issue because a secondary phone is working fine, my connection to > > the server is fine too, any ideas or suggestions are welcome. > > > > -Motty > > We use Polycom 450s as the main desk phone throughout our company and have > no issues with them registering. Without seeing your configs, it is hard to > give you specific advice. The part of your configs you need to be looking > at (assuming you are provisioning from http or ftp), are the following: > > For the Polycom look at the reg section: >reg.1.auth.userId="6534" reg.1.auth.password="myPassword"> > > For Asterisk (in your sip.conf or other appropriate config file): > > [6534](polycom) > callerid="Bob SMith" <6534> > secret=myPassword > mailbox=6534 > > A couple of notes here: 6534 is the extension number for Bob Smith. > "myPassword" in the files should be replaced with whatever password you > have assigned for that phone. The (polycom) template contains all the > options needed for phones to work in my specific install, but doesn't have > anything that would affect registration. > > If you watch the asterisk console when you boot up the phone, do you get > any errors in the console? I know when I am testing/experimenting with new > setups that I often see errors when the phone goes to register. It usually > is because I have either specified a username that doesn't exist in > Asterisk or I have the phone passing an incorrect password with what is > specified in sip.conf. > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk
I believe you're correct. And that should be the correct setting. However, You may want to do a packet sniff and confirm you're seeing the actual traffic as expected. Being that you see timeouts on the asterisk side. My bet is the rtp/sip traffic is going toward the device on a port it's not expecting. Or, The NAT device doesn't have a mapping for and being dropped at one of your routing devices. Nick Olsen Network Operations (855) FLSPEED x106 From: "John Millican" Sent: Thursday, January 02, 2014 11:07 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk top posting so as to not make thread even more confusing. Nick, I have nat=force_rport,comedia in sip.conf. It is my understanding that nat=yes is deprecated? Thanks, JohnM On 01/02/2014 10:51 AM, Nick Olsen wrote: > Make sure you have nat=yes in your sip.conf either under globals or > individual sip peer settings. > > Nick Olsen > Network Operations > (855) FLSPEED x106 > > > > > *From*: "John Millican" > *Sent*: Thursday, January 02, 2014 10:50 AM > *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" > > *Subject*: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> > NAT/Firewall-> Asterisk > > Hello, > CentOS 6.x and Asterisk 11.x > I have an interesting, to me at least, situation. Using a Polycom > 501(also tried with X-Lite). I have set up Asterisk to accept > registration from the Polycom and it registers successfully but then > withing 30 seconds on the CLI I get the message that the Polycom is > unreachable. The phone still shows that it is registered and if I try > to place a call from the phone to my Cell, my cell rings once and then > stops. I get a packet retransmission error: > WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout > reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical > Response) > Followed by: > n_sip.c:4203 retrans_pkt: Hanging up call xx@192.168.0.100 - no > reply to our critical packet > I am "assuming" that there is a problem with NAT. I have externip set > in sip.conf. > Any pointers to what I am missing? > Thanks, > JohnM > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450
asterisk-users-boun...@lists.digium.com wrote on 01/02/2014 10:03:19 AM: > From: motty cruz > To: Asterisk Users Mailing List - Non-Commercial Discussion > , > Date: 01/02/2014 10:02 AM > Subject: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450 > Sent by: asterisk-users-boun...@lists.digium.com > > Hello, I'm having issues with my phone Polycom sp450 not subscribing > to Asterisk server. Asterisk server is fine, firewall is not the > issue because a secondary phone is working fine, my connection to > the server is fine too, any ideas or suggestions are welcome. > > -Motty We use Polycom 450s as the main desk phone throughout our company and have no issues with them registering. Without seeing your configs, it is hard to give you specific advice. The part of your configs you need to be looking at (assuming you are provisioning from http or ftp), are the following: For the Polycom look at the reg section: For Asterisk (in your sip.conf or other appropriate config file): [6534](polycom) callerid="Bob SMith" <6534> secret=myPassword mailbox=6534 A couple of notes here: 6534 is the extension number for Bob Smith. "myPassword" in the files should be replaced with whatever password you have assigned for that phone. The (polycom) template contains all the options needed for phones to work in my specific install, but doesn't have anything that would affect registration. If you watch the asterisk console when you boot up the phone, do you get any errors in the console? I know when I am testing/experimenting with new setups that I often see errors when the phone goes to register. It usually is because I have either specified a username that doesn't exist in Asterisk or I have the phone passing an incorrect password with what is specified in sip.conf.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450
Which firmware version? 4.1.x is only for use with MS Link server. A symptom of running 4.1.x firmware with a non-MS server is the phone will not show buddies. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz Sent: Thursday, January 02, 2014 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450 Hello, I'm having issues with my phone Polycom sp450 not subscribing to Asterisk server. Asterisk server is fine, firewall is not the issue because a secondary phone is working fine, my connection to the server is fine too, any ideas or suggestions are welcome. -Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk
top posting so as to not make thread even more confusing. Nick, I have nat=force_rport,comedia in sip.conf. It is my understanding that nat=yes is deprecated? Thanks, JohnM On 01/02/2014 10:51 AM, Nick Olsen wrote: > Make sure you have nat=yes in your sip.conf either under globals or > individual sip peer settings. > > Nick Olsen > Network Operations > (855) FLSPEED x106 > > > > > *From*: "John Millican" > *Sent*: Thursday, January 02, 2014 10:50 AM > *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" > > *Subject*: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> > NAT/Firewall-> Asterisk > > Hello, > CentOS 6.x and Asterisk 11.x > I have an interesting, to me at least, situation. Using a Polycom > 501(also tried with X-Lite). I have set up Asterisk to accept > registration from the Polycom and it registers successfully but then > withing 30 seconds on the CLI I get the message that the Polycom is > unreachable. The phone still shows that it is registered and if I try > to place a call from the phone to my Cell, my cell rings once and then > stops. I get a packet retransmission error: > WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout > reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical > Response) > Followed by: > n_sip.c:4203 retrans_pkt: Hanging up call xx@192.168.0.100 - no > reply to our critical packet > I am "assuming" that there is a problem with NAT. I have externip set > in sip.conf. > Any pointers to what I am missing? > Thanks, > JohnM > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450
Hello, I'm having issues with my phone Polycom sp450 not subscribing to Asterisk server. Asterisk server is fine, firewall is not the issue because a secondary phone is working fine, my connection to the server is fine too, any ideas or suggestions are welcome. -Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get data from the SDPof SIP INVITE message
Mordechay Kaganer wrote: B.H. Hello, all I'm using Asterisk 11.7, connected to PSTN using SIP trunk. I'm looking for a way to get data from INVITE's SDP. Specifically, i would like to get a value of o= for incoming call from PSTN because it contains data about the operator that the call originates from. I'm afraid not, the only information in the dialplan even remotely relating to SDP is the RTP address information. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 12 trunk setup
Kilburn Abrahams wrote:
Hi All
Hola,
I am testing Asterisk 12 and got most things working, but cannot get a
trunk setup working. I am using the new pjsip channel driver. The
provider provides IP security so no registering or credentials are
required.
My version12
[udpnonat]
type=transport
protocol=udp
bind=0.0.0.0:5060
[maintrunk]
type=endpoint
transport=udpnonat
disallow=all
allow=g729,alaw,ulaw
aors=maintrunk
[maintrunk]
type=aor
contact=sip:1.2.3.4:5060
and use Dial(PJSIP/${ARG1}@maintrunk)
It dials but does not connect to the provider. Is the config correct?
Your config itself looks fine, what actually shows up on the console?
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk
Make sure you have nat=yes in your sip.conf either under globals or individual sip peer settings. Nick Olsen Network Operations (855) FLSPEED x106 From: "John Millican" Sent: Thursday, January 02, 2014 10:50 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk Hello, CentOS 6.x and Asterisk 11.x I have an interesting, to me at least, situation. Using a Polycom 501(also tried with X-Lite). I have set up Asterisk to accept registration from the Polycom and it registers successfully but then withing 30 seconds on the CLI I get the message that the Polycom is unreachable. The phone still shows that it is registered and if I try to place a call from the phone to my Cell, my cell rings once and then stops. I get a packet retransmission error: WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical Response) Followed by: n_sip.c:4203 retrans_pkt: Hanging up call xx@192.168.0.100 - no reply to our critical packet I am "assuming" that there is a problem with NAT. I have externip set in sip.conf. Any pointers to what I am missing? Thanks, JohnM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk
Hello, CentOS 6.x and Asterisk 11.x I have an interesting, to me at least, situation. Using a Polycom 501(also tried with X-Lite). I have set up Asterisk to accept registration from the Polycom and it registers successfully but then withing 30 seconds on the CLI I get the message that the Polycom is unreachable. The phone still shows that it is registered and if I try to place a call from the phone to my Cell, my cell rings once and then stops. I get a packet retransmission error: WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical Response) Followed by: n_sip.c:4203 retrans_pkt: Hanging up call xx@192.168.0.100 - no reply to our critical packet I am "assuming" that there is a problem with NAT. I have externip set in sip.conf. Any pointers to what I am missing? Thanks, JohnM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] screen capture for asterisk call center solution
On Friday 20 December 2013, Goke M Aruna wrote: > hello AJ, > Can I benefit from that code of yours? > regards As it was written for a very specific application, it's extremely unlikely that it will suit your environment. If you are wanting me to write some application-specific software for your business, then you will need to contact me separately, as this list is strictly for *non*-commercial discussion. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
