[asterisk-users] sip trunk, parsing DID

Hello, I am using a Swiss VoIP provider called sipcall. They have what they call a SIP trunk, and it is less expensive than individual accounts. From Asterisk's point of view, this is just a regular SIP account, which can however receive and send calls from multiple numbers. I just migrated from

Re: [asterisk-users] SIP trunk problem: Message option 200 (heartbeat)

On Wed, Sep 4, 2019, at 6:01 AM, bilal ghayyad wrote: > Hello; > > I am facing a trouble with the SIP service provider, they are saying > that there is a problem related to message option 200 (the heartbeat), > so what is required to add this for the sip configuration? Below is my > sip debug

[asterisk-users] SIP trunk problem: Message option 200 (heartbeat)

Hello; I am facing a trouble with the SIP service provider, they are saying that there is a problem related to message option 200 (the heartbeat), so what is required to add this for the sip configuration? Below is my sip debug trace log with the them and the sip peer configuration: [Sep  4

Re: [asterisk-users] SIP trunk between asterisk boxes

On Tue, Jul 23, 2019, at 2:53 PM, Jerry Geis wrote: > > rtp set debug on" will show the RTP traffic flowing,I did not see anything > > printed when I pressed a key. I say the audio prints. That means either it was not negotiated or was not picked up by Asterisk. -- Joshua C. Colp Digium - A

Re: [asterisk-users] SIP trunk between asterisk boxes

> rtp set debug on" will show the RTP traffic flowing, I did not see anything printed when I pressed a key. I say the audio prints. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

Re: [asterisk-users] SIP trunk between asterisk boxes

On Tue, Jul 23, 2019, at 1:47 PM, Jerry Geis wrote: > I have a sip trunk between two asterisk boxes. > I can call into the first box, hit 499 for example and the call goes to > the second box and answers as expected plays me audio message just fine > etc... My issue is that DTMF does not seem

[asterisk-users] SIP trunk between asterisk boxes

I have a sip trunk between two asterisk boxes. I can call into the first box, hit 499 for example and the call goes to the second box and answers as expected plays me audio message just fine etc... My issue is that DTMF does not seem to be working. Both sides are set for: dtmfmode=RFC2833 What

Re: [asterisk-users] sip trunk with social media

On Thursday 04 January 2018 at 01:27:59, bilal ghayyad wrote: > Hello > It will be amazing if possible to do sip trunk with any of social media > providers like: whatsapp, facebook, imo, viber, ... etc To the best of my knowledge none of the services you mention either operate over SIP or

[asterisk-users] sip trunk with social media

Hello It will be amazing if possible to do sip trunk with any of social media providers like: whatsapp, facebook, imo, viber, ... etc.Did anyone has luck with this? RegardsBilal Sent from Yahoo Mail on Android-- _ -- Bandwidth

Re: [asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)

> Not sure maybe there's a better solution but I thought about using another > peer with type=user for incoming connections. That's what I've done for my connection to the service provider I use (Vitelity), as they have different inbound and outbound hosts/proxies. This works fine. --

Re: [asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)

Not sure maybe there's a better solution but I thought about using another peer with type=user for incoming connections. On Mon, May 22, 2017, 6:13 PM Benoit Panizzon wrote: > Hello List > > I work at an SIP Provider and we have added and SBC in front of our > Voice

[asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register =>

Re: [asterisk-users] SIP trunk down. Wireshark shows ICMP Communication administratively filtered

Sounds like a firewall setting to me. If you can ping, then Internet Control Message Protocol (ICMP) packets are allowed, but if SIP traffic is returning the ICMP Type 3 (code 13) response, then your SIP ports are blocked (at least the firewall admin was nice enough to leave the reason code

[asterisk-users] SIP trunk down. Wireshark shows ICMP Communication administratively filtered

Hello, I've got a remote system that is plagued with a strange issue. It happens from time to time. Yet, I've not found any condition that trigger this phenomenon. Here is my setup: - PSTN <---> ITSP <--SIP trunk--> Router <> Switch <> Asterisk box | |

Re: [asterisk-users] SIP trunk

On Tuesday 26 Jul 2016, Jerry Geis wrote: > It seems I am not getting any digits coming over a SIP trunk. > > How can I match "anything" or "nothing" and start my extension. > > Usually I have something like: > exten => 55,1,Goto(,yyy,1) > > but if 55 does not come across and it appears to

Re: [asterisk-users] SIP trunk

Hi Jerry, In article , Jerry Geis wrote: > > It seems I am not getting any digits coming over a SIP trunk. > > How can I match "anything" or "nothing" and start my extension. > > Usually I have

Re: [asterisk-users] SIP trunk

_. ? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On Tue, Jul 26, 2016 at 11:39 AM, Jerry Geis wrote: > It seems I am not

[asterisk-users] SIP trunk

It seems I am not getting any digits coming over a SIP trunk. How can I match "anything" or "nothing" and start my extension. Usually I have something like: exten => 55,1,Goto(,yyy,1) but if 55 does not come across and it appears to be no digits coming across how do I match that that and

Re: [asterisk-users] SIP trunk with whatsapp

On 03/29/16 17:03, Vitor Mazuco wrote: > Is possible with Telegram? Telegram does not support voice calls for humans either. It's strictly an IM system. They do have a bot API if you want to interface some system with their messaging system. With it you can send text and also pictures,

Re: [asterisk-users] SIP trunk with whatsapp

Is possible with Telegram? 2016-03-29 9:39 GMT-03:00, Emiliano Vazquez : > El 29/03/16 a las 08:29, Steve Howes escribió: >> I don't think you can. Whatsapp is a closed system. >> >> Steve > And they change your code every day and make it always obfuscated. > >

Re: [asterisk-users] SIP trunk with whatsapp

El 29/03/16 a las 08:29, Steve Howes escribió: I don't think you can. Whatsapp is a closed system. Steve And they change your code every day and make it always obfuscated. https://github.com/tgalal/yowsup/issues/887 Best regards. Emiliano. --

Re: [asterisk-users] SIP trunk with whatsapp

On 28/03/16 12:46, bilal ghayyad wrote: Does anyone has information if possible to setup SIP trunk with whatsapp? How can we let asterisk send and receive calls from whatsapp? I don't think you can. Whatsapp is a closed system. Steve --

[asterisk-users] SIP trunk with whatsapp

Hello; Does anyone has information if possible to setup SIP trunk with whatsapp? How can we let asterisk send and receive calls from whatsapp? RegardsBilal-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] SIP Trunk - problem to connect

Hello! Thnxs for reading! I've an IPLAN virtual PBX, that allows me to connect via zoiper or gigaset, for instance (and it works!) Connection parameters are: Authentication Name: Número 11 Authentication password: 12345678 Username: 11 Display name: 11 Domain:

[asterisk-users] sip trunk to Cisco router

hello everybody, i want to configure a sip trunk between my system which has asterisk 11.5.1 and a cisco router. this is my scenario: Freepbx-my system-cisco-routerFreepbx my system acts like a router. if i set just one codec in dial-peers on cisco router, every thing is ok and i

[asterisk-users] SIP trunk no audio

I have two machines on the internet. Box A and Box B. Box A has a SIP trunk to the world, Making calls box A works fine I have audio to my cell and all works. I defined a SIP trunk between box B and A. tried to make a call originating from box B - to box A and then over the SIP trunk to my cell.

Re: [asterisk-users] SIP trunk no audio

But the phone rings - so its routed - just no audio. The ringing is SIP signaling. The audio is RTP data. See if the audio is getting routed with a sniffer. Maybe use one codec that both clients support. Adrian Serafini --

Re: [asterisk-users] SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

Is the quality the same incoming from mobile as outgoing to mobile? On Wed, Jul 30, 2014 at 4:51 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio

Re: [asterisk-users] SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

On Thursday 31 Jul 2014, James Thomas wrote: Is the quality the same incoming from mobile as outgoing to mobile? It's a one-way trunk (outgoing only). Anyway, I've now fixed it, with help from the trunk provider. Details to follow in a separate message. -- AJS Note: Originating address

[asterisk-users] SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio quality. I thought it might have been a codec issue; we have used G.726 for internal and external calls (over primary ISDN and GSM). So I tried allowing

Re: [asterisk-users] SIP trunk and congestion handling

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mordechay Kaganer Sent: Thursday, August 15, 2013 5:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP trunk and congestion handling B.H. While dialing out i get a lot of AMI responses like

Re: [asterisk-users] SIP trunk and congestion handling

:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mordechay Kaganer *Sent:* Tuesday, August 13, 2013 10:55 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] SIP trunk and congestion handling

Re: [asterisk-users] SIP trunk and congestion handling

...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mordechay Kaganer *Sent:* Tuesday, August 13, 2013 10:55 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] SIP trunk and congestion handling ** ** B.H. Asterisk

Re: [asterisk-users] SIP trunk and congestion handling

Kaganer *Sent:* Sunday, August 11, 2013 8:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] SIP trunk and congestion handling ** ** B.H. ** ** Hello, all. We have a dialer software that runs outgoing telephony campaigns. We have been

Re: [asterisk-users] SIP trunk and congestion handling

-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mordechay Kaganer Sent: Tuesday, August 13, 2013 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP trunk and congestion handling B.H. Asterisk 1.8.22

Re: [asterisk-users] SIP trunk and congestion handling

Which version of asterisk are you using ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mordechay Kaganer Sent: Sunday, August 11, 2013 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP

[asterisk-users] SIP trunk and congestion handling

B.H. Hello, all. We have a dialer software that runs outgoing telephony campaigns. We have been using it successfully with PRI cards, now we're evaluating it's use also with a SIP trunk. Most of the things run perfectly good without a need to change anything except for dial string, but there's

Re: [asterisk-users] sip trunk failing to register causes sip phones to become unreachable

Hi, Thanks. I will try this. Regards, John - Original Message - From: Warren Selby wcse...@selbytech.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, 23 August, 2012 9:24:48 PM Subject: Re: [asterisk-users] sip trunk

[asterisk-users] sip trunk failing to register causes sip phones to become unreachable

Hi, I have only seen this problem when using sipgate SIP trunks which actually register. If the ADSL connection goes down that the sip trunk uses, the sip phones registered locally become unreachable. This happens on any 1.6.x or 1.8 version of asterisk I've tried. Is there a work around that

Re: [asterisk-users] sip trunk failing to register causes sip phones to become unreachable

On Aug 23, 2012, at 10:30 AM, John Cahill j...@dmcip.com wrote: I have only seen this problem when using sipgate SIP trunks which actually register. If the ADSL connection goes down that the sip trunk uses, the sip phones registered locally become unreachable. This happens on any 1.6.x or

Re: [asterisk-users] sip trunk failing to register causes sip phones to become unreachable

To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip trunk failing to register causes sip phones to become unreachable On Aug 23, 2012, at 10:30 AM, John Cahill j...@dmcip.com wrote: I have only seen this problem when using sipgate SIP trunks which actually

[asterisk-users] SIP trunk audio bad but is OK again after SIP re-registration

Hi, When *ANY* SIP client (softphone, hardphone, ATA) registers to an Asterisk server on my LAN and the extension dials out through a remote SIP provider, the audio is fine for a while. It then degrades and starts to be cracky/jittery. The extension can call once and again and it will always

[asterisk-users] SIP trunk call initiated as Anonymous@anonymous.invalid

I have a Grandstream HT-502 device connected to my Asterisk PBX. It is configured not to place anonymous calls, and from my mostly layman reading of the invitation that the device sends, it should not be anonymous. However, the Asterisk PBX sends an anonymous invitation to our SIP trunk

[asterisk-users] SIP Trunk

Hi, I have a situation where unfortunately, I cannot use IAX for trunks, and need to instead use SIP trunks. Is there any way to fit the voice data from more than one simultaneous phone call into a single IP packet over the SIP trunk. I believe this is possible with IAX trunks, but I don't know

Re: [asterisk-users] SIP Trunk

No. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Courtier-Dutton Sent: Friday, December 16, 2011 5:30 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP Trunk Hi, I have

Re: [asterisk-users] SIP Trunk

16 dec 2011 kl. 11:29 skrev James Courtier-Dutton: Hi, I have a situation where unfortunately, I cannot use IAX for trunks, and need to instead use SIP trunks. Is there any way to fit the voice data from more than one simultaneous phone call into a single IP packet over the SIP trunk. I

[asterisk-users] SIP trunk trouble. Please help.

Hello, I am troubleshooting a SIP trunk problem. The system is Asterisk 1.8.5. The problem is can't make any outbound/inbound. It always get Number is not valid 701. I tried to figure out the reason the call got dropped and couldn't find out the solution. I noticed that in the SIP debug

[asterisk-users] sip trunk balancing

hi, is there some way to balance accross sip trunks by the number of calls? example 3 trunks alfa(max 30 channels, priority 1),delta(60,priority 2),omega(90,priority 3) alfa have 25 calls now i want next call terminate to delta. how to find in asterisk the current calls number on sip trunk

Re: [asterisk-users] sip trunk balancing

On 02/03/2011 11:41 AM, marek cervenka wrote: hi, is there some way to balance accross sip trunks by the number of calls? example 3 trunks alfa(max 30 channels, priority 1),delta(60,priority 2),omega(90,priority 3) alfa have 25 calls now i want next call terminate to delta. how to find in

Re: [asterisk-users] SIP Trunk configuration problem - fromdomain

you need to set your external IP in the sip.conf to be your public IP after NAT (assuming your talking over a public network). That way when the sip request goes out and it sees the IP address your are sending to is outside your localnets it changes the SIP header to use the x.y.z.w IP you set

[asterisk-users] SIP Trunk configuration problem - fromdomain

Hello, I'm trying to register to my provider sip trunk, I got from him an host IP (a.b.c.d) to connect to and my provider recognize me based on the fixed IP (x.y.z.w) he gave me (no need for username and password) In the sip.conf I add: [mytrunk] type=friend insecure=no host=a.b.c.d

Re: [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]

Hi! Issue solved. Looks like all I was missing was one parameter: fromuser= That's interesting - could be related to this: http://lists.digium.com/pipermail/asterisk-dev/2006-November/024842.html You were probably caught be the fact that you are using extension numbers also as SIP user

Re: [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]

--- On Fri, 5/14/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: You were probably caught be the fact that you are using extension numbers also as SIP user names for your phones (here: 3666). This is not a good thing to do, better use an alphanumeric username or

Re: [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]

Issue solved. Looks like all I was missing was one parameter: fromuser= Thanks for your time! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

[asterisk-users] SIP trunk between two Asterisk servers

Hi, I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls). With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious. I followed the simple example at http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/. so

Re: [asterisk-users] SIP trunk between two Asterisk servers

Hi! I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls). With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious. Either a) set a secret and use that on both sides, or b) look at allowguest= and the default

Re: [asterisk-users] SIP trunk between two Asterisk servers

--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Either a) set a secret and use that on both sides, or b) look at allowguest= and the default context and maybe the domain= settings, or c) use insecure=invite Thanks Philipp. I'm trying option

Re: [asterisk-users] SIP trunk between two Asterisk servers

Hello Server1: sip.conf [interboxserver2] type=friend host=192.168.250.112 context=callfromserver2 disallow=all allow=ulaw allow=alaw allow=g729 extensions.conf [callfromserver2] exten = _X.,1,Noop(Call from server2) exten = _X.,2,Dial(SIP/${EXTEN}) exten = _X.,3,Hangup Server2: sip.conf

Re: [asterisk-users] SIP trunk between two Asterisk servers

Vardan wrote: Hello Server1: sip.conf [interboxserver2] type=friend host=192.168.250.112 context=callfromserver2 disallow=all allow=ulaw allow=alaw allow=g729 extensions.conf [callfromserver2] exten = _X.,1,Noop(Call from server2) exten = _X.,2,Dial(SIP/${EXTEN}) exten =

Re: [asterisk-users] SIP trunk between two Asterisk servers

And also please show your settings and logs (without debug) Vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: Either a) set a secret and use that on both sides, or b) look at allowguest= and the default context and maybe the

Re: [asterisk-users] SIP trunk between two Asterisk servers

I have forget to write for outcall in extension server1: [calltoserver2] exten = _X.,1,Noop(Call to server2) exten = _X.,2,Dial(SIP/interboxserver2/${EXTEN}) exten = _X.,3,Hangup server2: [calltoserver1] exten = _X.,1,Noop(Call to server1) exten =

Re: [asterisk-users] SIP trunk between two Asterisk servers

Hi! I'm trying option c) which is the simplest. used insecure=invite but failed with the same SIP messages. Tried also insecure=yes but the same messages show up: SIP/2.0 407 Proxy Authentication Required Then you have another entry in sip.conf that uses the same IP address. Delete that,

Re: [asterisk-users] SIP trunk between two Asterisk servers

Hi again! --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after

Re: [asterisk-users] SIP trunk between two Asterisk servers

--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: I have forget to write for outcall in extension server1: [calltoserver2]   exten =  _X.,1,Noop(Call to server2)   exten =  _X.,2,Dial(SIP/interboxserver2/${EXTEN})   exten =  _X.,3,Hangup server2: [calltoserver1]   exten = 

Re: [asterisk-users] SIP trunk between two Asterisk servers

--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look

Re: [asterisk-users] SIP trunk between two Asterisk servers

--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: SIP/2.0 407 Proxy Authentication Required Then you have another entry in sip.conf that uses the same IP address. Delete that, or change the port on one of them, and adjust insecure= accordingly.

Re: [asterisk-users] SIP trunk between two Asterisk servers

Please look in any conf file that have any relations with sip.conf. I think you have some records. And one also, you take this message when calling in both direction? (server1 call server2 and server2 call server1) Vardan Vieri wrote: --- On Wed, 5/12/10, Vardanhvarda...@gmail.com wrote:

Re: [asterisk-users] SIP trunk between two Asterisk servers

please show sip show users and sip show peers vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on

Re: [asterisk-users] SIP trunk between two Asterisk servers

And sip show registry Vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about

Re: [asterisk-users] SIP trunk between two Asterisk servers

--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: please show sip show users and sip show peers SERVER 2: sip show users (trimmed to just my sip test trunk): Username Secret Accountcode Def.Context ACL NAT interboxsip

Re: [asterisk-users] SIP trunk between two Asterisk servers

--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: And sip show registry sip show registry doesn't list anything regarding my interboxsip test trunk because I'm trying to setup a straightforward link such as this one described here (without user/password):

Re: [asterisk-users] SIP trunk between two Asterisk servers

Please change the peers name in any server. for example: server1: interboxsip1 server2: interboxsip2 Vardan Vieri wrote: --- On Wed, 5/12/10, Vardanhvarda...@gmail.com wrote: please show sip show users and sip show peers SERVER 2: sip show users (trimmed to just my sip test trunk):

Re: [asterisk-users] SIP trunk between two Asterisk servers

What are your allowguest= and domain= settings in the global section of sip.conf? And which version of Asterisk exactly are you using? Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] SIP trunk between two Asterisk servers

--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: What are your allowguest= and domain= settings in the global section of sip.conf? And which version of Asterisk exactly are you using? I have no such settings defined yet. Still haven't tried to

Re: [asterisk-users] SIP trunk between two Asterisk servers

--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: Please change the peers name in any server. for example: server1: interboxsip1 server2: interboxsip2 If I understand correctly, the peer names can be identical on both servers. What counts is the host entry, I guess. But then

Re: [asterisk-users] SIP Trunk with multiple remote ip-addresses

Am 02.03.2010 13:29, schrieb Magnus Benngård: Hi! Did a setup of 2 peers as Klaus suggested, it worked thx! Has anyone thought about the possibility to add multiple ip/hosts to host=? I my case: host=130.244.190.42,130.244.190.46 or host=sip-corporate1.tele2.se,sip-corporate2.tele2.se

Re: [asterisk-users] SIP Trunk with multiple remote ip-addresses

12 mar 2010 kl. 12.01 skrev Klaus Darilion: Am 02.03.2010 13:29, schrieb Magnus Benngård: Hi! Did a setup of 2 peers as Klaus suggested, it worked thx! Has anyone thought about the possibility to add multiple ip/hosts to host=? I my case: host=130.244.190.42,130.244.190.46 or

Re: [asterisk-users] SIP Trunk with multiple remote ip-addresses

Am 02.03.2010 08:50, schrieb Magnus Benngård: Hi, Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No problem to get outgoing calls to work but i have some problems with incoming. Did set srvlookup=yes in sip.conf. Sending all outgoing calls to sip-corporate.tele2.se which

Re: [asterisk-users] SIP Trunk with multiple remote ip-addresses

Hi! Did a setup of 2 peers as Klaus suggested, it worked thx! Has anyone thought about the possibility to add multiple ip/hosts to host=? I my case: host=130.244.190.42,130.244.190.46 or host=sip-corporate1.tele2.se,sip-corporate2.tele2.se Step 1 could be to send to the first ip/host and

[asterisk-users] SIP Trunk with multiple remote ip-addresses

Hi, Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No problem to get outgoing calls to work but i have some problems with incoming. Did set srvlookup=yes in sip.conf. Sending all outgoing calls to sip-corporate.tele2.se which is either sip-corporate1.tele2.se (130.244.190.42)

[asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops

Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines

Re: [asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops

do you use the qualify=yes option for your endpoints? y. Peter Childs schrieb: Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing

Re: [asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops

2010/1/26 Yves Arikoglu yves...@gmx.de: do you use the qualify=yes No, If I do it does not work at all. I've found if I set defaultexpiry to 30 it works fine. and was infact working for 30 seconds every two minutes before, It looks like sipgate.co.uk are expiring there registry attempts very

Re: [asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops

2010/1/26 Peter Childs pchi...@bcs.org: 2010/1/26 Yves Arikoglu yves...@gmx.de: do you use the qualify=yes No, If I do it does not work at all. I've found if I set defaultexpiry to 30 it works fine. and was infact working for 30 seconds every two minutes before, It looks like

[asterisk-users] sip trunk that fails over time

Hi, I have configure a SIP trunk between two asterisk 1.4.24.1 After a while, sometimes a day or two, sometimes only a few hours, the SIP connection between the two servers is lost. 'sip show peer status' shows the peer is unreachable. 'sip reload' resolves the problem, but I'm wondering if

Re: [asterisk-users] sip trunk that fails over time

Hi Have you tried setting qualify in the sip.conf? http://www.voip-info.org/wiki/view/Asterisk+sip+qualify Ish dan julius wrote: Hi, I have configure a SIP trunk between two asterisk 1.4.24.1 After a while, sometimes a day or two, sometimes only a few hours, the SIP connection between

Re: [asterisk-users] sip trunk that fails over time

Yes, I have qualify=yes Could this be related to various posts regarding DNS issues? I doubt I have dns issues because the hostname and IP of the other server is hard-coded in /etc/hosts Thanks, Dan On Tue, Jul 28, 2009 at 3:38 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Have you

Re: [asterisk-users] SIP Trunk groups

, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: mlecu...@gmail.com Date: Wed, 27 May 2009 14:17:23 -0300 To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP Trunk groups Hey all, I have 2 GSM to Voip gateways and probably we will grow up to 4 more gateways. I

[asterisk-users] SIP Trunk groups

Hey all, I have 2 GSM to Voip gateways and probably we will grow up to 4 more gateways. I already created a macro to make failover happen between gateways, but can imagine that everytime I add a new gateway I will need to modify the macro. The initial intention of this macro was to failover

Re: [asterisk-users] SIP Trunk groups

AFAIK, unfortunatelly it's not the same as with ZAP channels where you can group multiple lines together. I ended up using slightly modified superdial macro: http://www.voip-info.org/wiki/view/Superdial+macro. if you add new gateway it's not necesarry to edit the macro, just add new line in

Re: [asterisk-users] SIP Trunk groups

I've improved this since this revision, but now a days I don't use limited systems. But my code has been used in places that need 100 concurrent outgoing lines. [macro-which-line] exten = s,1,set(TRIES=0) exten = s,n(nextone),set(TRIES=$[${TRIES} + 1]) ; increment TRIES by 1 exten =

[asterisk-users] SIP Trunk groups

He all, I have 2 GSM to Voip gateways and probably we will grow up to 4 more gateways. I already created a macro to make failover happen between gateways, but can imagine that everytime I add a new gateway I will need to modify the macro. The initial intention of this macro was to failover

Re: [asterisk-users] SIP trunk with 250 lines

Danny Nicholas da...@debsinc.com writes: Okay - I'm not shooting from the hip here. The driver in question is a Intel E1000 on a Poweredge 1650. If you visit the Digium site and do other googling, you will see that there is a specific issue with asterisk and this hardware/driver

Re: [asterisk-users] SIP trunk with 250 lines

@lists.digium.com Subject: Re: [asterisk-users] SIP trunk with 250 lines Danny Nicholas da...@debsinc.com writes: Okay - I'm not shooting from the hip here. The driver in question is a Intel E1000 on a Poweredge 1650. If you visit the Digium site and do other googling, you will see

[asterisk-users] SIP trunk with 250 lines

Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will be connected to the server through the same SIP trunk as

Re: [asterisk-users] SIP trunk with 250 lines

...@lists.digium.com] On Behalf Of Christian Victor Sent: Tuesday, March 24, 2009 10:10 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP trunk with 250 lines Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards

Re: [asterisk-users] SIP trunk with 250 lines

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Tuesday, March 24, 2009 10:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SIP trunk with 250 lines First Issue to be addressed is how

Re: [asterisk-users] SIP trunk with 250 lines

2009/3/24 Christian Victor christ...@victormedia.de Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will

Re: [asterisk-users] SIP trunk with 250 lines

2009/3/24 Cary Fitch ca...@usawide.net First Issue to be addressed is how many simultaneous calls and bandwidth availability. Number of “lines” (numbers) is not a limitation in it self unless they are all in use. Sorry for being a bit unclear in this point. What I meant was 240 to 480

Re: [asterisk-users] SIP trunk with 250 lines

2009/3/24 Danny Nicholas da...@debsinc.com Here are a few “look outs”; Using conference rooms will increase your bandwidth requirements. On board Network controllers will affect performance in this “high-use” scenario. 250 simultaneous calls will use about 7.5Mb of bandwidth depending on

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